In this paper, we study the feature of Multi-Flow HSDPA (MF-HSDPA), which allows simultaneous transmission of separate data packets from two or more serving cells (across different sectors or NodeBs) to a UE. This new feature brings challenges for system design, especially for the Inter-NodeB case, in which the serving cells reside in different NodeBs. To address these challenges, we propose enhancements to the Radio Link Control (RLC) layer and Media Access Control (MAC) layer. Simulation results show significant performance gain can be achieved by the MF-HSDPA feature.

In heterogeneous networks, low power nodes may be deployed within macro cells for coverage and capacity improvement. Due to transmit power difference between low power nodes and macro cells, downlink (DL) and uplink (UL) coverage of the low power nodes can be very different. This DL-UL coverage difference could negatively affect system performance on both the DL and the UL. We describe a few techniques to alleviate this problem in HSPA. We present simulation results to show the effectiveness of these techniques as well as the system performance improvement of heterogeneous networks over traditional macro cell only deployments in HSPA.

Heterogeneous networks (HetNets), consisting of Macro NodeBs (macros) and low power Pico NodeBs (picos), can increase system capacity by offloading some users to the picos. However, this offloading may be limited due to intercell interference from macros to picos that reduces their coverage. HSPA Multi-carrier HetNets allow the use of range expansion techniques to achieve more offloading, such as power reduction from the macros on one carrier. This paper provides a system level analysis of the gain provided by power reduction on one carrier of the macros in a HetNet environment. As the interference from the macros is reduced, more capacity offloading to picos occurs and higher system capacity can be achieved. By considering cell biasing in serving cell selection, further gains can be observed.

In this paper, we study the feature of Multi-Flow HSDPA (MF-HSDPA), which allows simultaneous transmission of separate data packets from two or more serving cells (across different sectors or NodeBs) to a User Equipment (UE). This new feature brings challenges for system design, especially for the Inter-NodeB case, in which the serving cells reside in different NodeBs. To address these challenges, we propose enhancements to the Radio Link Control (RLC) layer and backhaul flow control/congestion control. Simulation results show significant performance gain can be achieved by the MF-HSDPA feature. The gain is more promising when the backhaul is congested, because Inter-NodeB MF-HSDPA offers "backhaul diversity" by utilizing two backhaul links to serve a MF UE.

In this paper, we analyze the capacity of Circuit Switched Voice over High Speed Packet Access (HSPA) (CSoHS), using a Release 7 HSPA system that supports Continuous Packet Connectivity (CPC). On the Downlink (DL), we modeled the power consumption of the Fractional Dedicated Physical Channel (F-DPCH) and the Enhanced DCH Hybrid ARQ Indicator Channel (E-HICH). On the uplink (UL), we modeled the overhead channels, High Speed Dedicated Physical Control Channel (HS-DPCCH) and Enhanced DCH Dedicated Physical Control Channel (E-DPCCH). Our results show voice capacity gains that are achievable with CPC. We also show the gains in downlink voice capacity that are achievable by using the MAC-segmentation feature, introduced in Release 7.

Dual Cell HSDPA (DC-HSDPA), which was introduced in Release 8 of the WCDMA specifications, enables the User Equipment (UE) to receive downlink data on two adjacent carriers simultaneously. Although the user physical layer throughput is expected to double compared to a single carrier HSDPA (SC-HSDPA) system, it is not quite clear how this would translate to user experience for realistic applications in various operating scenarios, i.e. channel conditions and loading. We implemented DC- HSDPA functionality in a prototype and evaluated its performance for various applications in comparison with SC-HSDPA. Our results show that, with DC-HSDPA, realistic applications such as web browsing and video streaming see significant gains in lab as well as OTA environments compared to SC-HSDPA with or without loading due to other users. Our results also highlight the ability of DC-HSDPA UEs to improve application performance by taking advantage of uneven loading across carriers, which may result due to UEs operating on a single carrier.

Dual Cell-High Speed Downlink Packet Access (DC-HSDPA), which was introduced in Release 8 of the WCDMA specifications, enables the User Equipment (UE) to receive downlink data on two adjacent carriers simultaneously. Previous papers, including [1] [2] have shown how the doubling of physical layer rate of DC-HSDPA UEs translates to user experience gain for real applications such as web browsing and video streaming in unloaded systems. In this paper, using a real prototype implementation, we focus on gains of DC-HSDPA over two carriers of Single Cell HSDPA (SC-HSDPA) in terms of number of additional supportable users. For a given user experience, we show how many more users DC-HSDPA can support compared to two carriers of SC-HSDPA; we show these results for the two applications, file download and web browsing. In addition, our results show the impact of the TCP Slow Start algorithm on the user experience gains of DC-HSDPA for the application of file download.

Legacy High Speed Downlink Shared Channel (HS-DSCH) Serving Cell Change (SCC) procedures in deployed HSPA networks (3GPP Release 5 and 6) may lead to unacceptably high call drop rates in realistic Urban Canyon conditions when Signaling Radio Bearers (SRBs) are mapped on the HS-DSCH. Mapping SRB on HS-DSCH is necessary to enable high capacity voice over HSPA (Circuit Switched Voice or VoIP over HSPA) since it eliminates the code limitation that arises when SRBs are mapped on DCH. To address the above robustness issue, Enhanced SCC (ESCC) was introduced in 3GPP Release 8. In this paper we highlight the first ever implementation of the E-SCC algorithm in a HSPA prototype system. A comprehensive evaluation of the E-SCC algorithm for voice over HSPA is performed through measurements made over a prototype network. We show that under tough urban canyon conditions, the ESCC procedure provides significant gains compared to the legacy USCC procedure in terms of call drop rates, packet drops and duration of serving cell change, thus allowing robust high quality voice service under mobility conditions. To conclude, this paper provides evidence that E-SCC algorithm implementation is feasible and when implemented, robust mobility for voice over HSPA can be achieved.

Circuit-Switched (CS) Voice Services over HSPA (CSoHS) was recently introduced for 3GPP WCDMA Release 7/8 systems. The benefits of this feature include improving voice and data system capacity by utilizing the enhancements offered by the shared packet transport of HSPA air interface while preserving the already widely deployed core networks. In this paper, we discuss the operation and implementation of CSoHS and analyze its performance via simulations. We show that the system capacity of CSoHS is significantly higher than Release 99 CS voice under similar system conditions and voice quality.

TCP throughput limitations over wireless links have received considerable attention in the last few years. One of the problems is that TCP congestion control interprets packet losses as an indication of congestion, whereas in wireless links, losses could be due to transient link quality degradations. In this paper, we propose and study a link layer solution and evaluate its effects on TCP in the context of Bluetooth. We enhance the Bluetooth link layer to make use of channel state information and accordingly adapt the Bluetooth packet type to enhance TCP throughput We propose a simple analytical method to determine the optimal packet type for a given channel state by adding FEC support or changing packet size. Since wireless interfaces, such as 802.11 or Bluetooth, can provide information regarding the channel state using relevant APIs, this simple enhancement can easily be added to the link layer. We implemented this functionality in the Bluetooth link layer. Our simulation experiments show that the proposed adaptive packet type solution significantly improves TCP throughput. The throughput enhancement increases with the error rate. For high error rates close to 0.1%, the link layer enhanced with the adaptive scheme is able to maintain good TCP throughput, whereas throughput is almost zero when the adaptive scheme is not used.

CapProbe is an inexpensive and accurate means to estimate capacity. CapProbe combines both dispersion and end-to-end delay to estimate the capacity of the narrowest link on a path. We evaluate in this paper the accuracy of CapProbe estimation, and its dependence on end systems speed, packet sizes, narrow link speeds, and other system parameters. We test kernel and user level implementations of CapProbe and find the kernel implementation to be much more accurate. We also evaluate through experiments the effect of probing packet size on the accuracy of CapProbe estimation. Finally, we explore the idea of a "passive CapProbe" within the context of a TCP flow. Passive here means that the dispersion and delay observed for the TCP flow data and ACK packets, without introducing any additional probing packets. We test active and passive versions of CapProbe with TCP. The active version is found to produce more accurate capacity estimates than the passive version.

We present a new capacity estimation technique, called CapProbe. CapProbe combines delay as well as dispersion measurements of packet pairs to filter out samples distorted by cross-traffic. CapProbe algorithms include convergence tests and convergence speed-up techniques by varying probing parameters. Our study of CapProbe includes a probability analysis to determine the time it takes CapProbe to converge on the average. Through simulations and measurements, we found CapProbe to be quick and accurate across a wide range of traffic scenarios. We also compared CapProbe with two previous well-known techniques, pathchar and pathrate. We found CapProbe to be much more accurate than pathchar and similar in accuracy to pathrate, while providing faster estimation than both. Another advantage of CapProbe is its lower computation cost, since no statistical post processing of probing data is required.

The problem of estimating the capacity of an Internet path is one of fundamental importance. Due to the multitude of potential applications, a large number of solutions have been proposed and evaluated. The proposed solutions so far have been successful in partially addressing the problem, but have suffered from being slow, obtrusive or inaccurate. In this work, we evaluate CapProbe, a low-cost and accurate end-to-end capacity estimation scheme that relies on packet dispersion techniques as well as end-to-end delays. The key observation that enabled the development of CapProbe is that both compression and expansion of packet pair dispersion are the result of queuing due to cross-traffic. By filtering out queuing effects from packet pair samples, CapProbe is able to estimate capacity accurately in most environments, with minimal processing and probing traffic overhead. In fact, the storage and processing requirements of CapProbe are orders of magnitude smaller than most of the previously proposed schemes. We tested CapProbe through simulation, Internet, Internet2 and wireless experiments. We found that CapProbe error percentage in capacity estimation was within 10% in almost all cases, and within 5% in most cases.

Bluetooth "scatternets" are targeting wider geographical area applications in factories, warehouses, shopping malls and various sensor network applications. Though some earlier work has looked at scatternet formation and scheduling issues, less attention has been given to optimizing scatternet topologies. We consider in this paper topological design of scatternets, taking into consideration application traffic requirements. We study appropriate topologies, and size the network in terms of piconets. We develop a scatternet queuing model and use it to compare the delay-throughput characteristics of various topologies. We find that the best topology is application independent. The analytical model is also used to determine the optimal point to operate a scatternet, i.e., the traffic load that saturates the network. We validate all out analytical results by simulations.

Streaming video as a form of media is becoming increasingly popular on the Internet. Real-time media such as video requires delay constraints from the network to ensure good quality at the receiver. While watching a video stream on his portable device connected to the Internet through the last-hop wireless link, the mobile user of tomorrow will expect a good experience. But, the time-varying nature of the wireless link can cause video frames to be dropped/delayed, affecting the quality of video at the receiver. In this paper, we propose a link layer approach to improve the quality of MPEG video streaming over a wireless link. We use Bluetooth as the wireless technology on which to test our scheme. Our results show that the quality of streaming video can be substantially improved with our scheme, particularly in bad channel conditions.

Bluetooth is a low-cost, low-power technology initially intended as a replacement of cables between electronic devices. Bluettoth devices can form small network of up to 8 devices called piconets. The specification also defines networks of piconets called scatternets. Scatternets can have various uses such as for "monitoring" purposes in factories and warehouses or for ad hoc meetings. Scatternets require the use of a routing scheme to find paths in a dynamic network. Though considerable research has been done in the area of routing in ad hoc networks, the direct application of this may be inefficient to Bluetooth scatternets. Some previous work has also presented routing schemes for scatternets, but this does not present any results showing the performance of the routing scheme. In this paper, we present a routing scheme for Bluetooth scatternets, which is based on the zone routing protocol. We motivate the design of the routing scheme keeping in mind the specifics of the Bluetooth technology. We present simulation results for the scheme, which show that the scheme gives very low overhead while keeping the route acquisition latencies low. The routing information at a node does not require a large amount of storage. In fact, a parameter in the scheme can be varied to trade-off storage information and routing overhead versus route acquisition latency.

Bluetooth scatternets are formed by interconnecting Bluetooth piconets in an ad-hoc fashion and consists typically of the handheld electronic devices from one or more user(s). Thus, scatternets may be used to form and interconnect ad-hoc Bluetooth personal area networks (PANs). The inter-piconet Bluetooth units, i.e. the gateways interconnecting the piconets in a scatternet, need to time division multiplex their presence in each of their piconets. This requires an inter-piconet scheduling (IPS) algorithm, operating in each inter-piconet unit in order to efficiently coordinate its presence with other Bluetooth units. In this paper an IPS algorithm based on periodic rendezvous points is proposed and analyzed with simulations, The algorithm is called maximum distance rendezvous point (MDRP) and utilizes the Bluetooth SNIFF functionality to establish the periodic rendezvous points between gateways and their peer nodes. Simulation results show that TCP works well with MDRP, but the large round trip delays caused by the inter-piconet gateway nodes give a slowly growing flow control window for TCP. The latter will in particular have an impact on the TCP performance in scatternets hosting "thin" (embedded) clients with limited dynamic memory capacity.

The standards that are supposed to play a leading role in third-generation mobile telecommunication and personal area networks in the near future are the Universal Mobile Telecommunication System (UMTS) and the Bluetooth (BT) radio technology, respectively. In this paper, we advocate that a hybrid architecture of UMTS and Bluetooth can take advantage of the complementary characteristics of these two technologies and provide a total solution for an indoor communication environment. We envision a cooperating scenario in which small Bluetooth networks (scatternets) offer basic wireless connectivity to several peripheral units scattered over small areas, while indoor UMTS supports communication among scatternets and provides wireless access to the Internet. We focus our analysis on a centralized topology, in which communication occurs only between the peripheral units and the access point. This topology can be used in many different application scenarios and represents an example of cooperation between 3G and PAN technologies. In addition to describing the architecture, we address the issue of fair capacity allocation in such a centralized topology and provide some analytic and simulation results for the topology considered.

Bluetooth is a new low-cost wireless technology that is going to play an important role in communications among small electronic devices and the access to wired networking infrastructure. Bluetooth stations that communicate directly form a piconet. In a piconet one station has the role of master and the others are slaves. The access to the medium is based on a TDD (time division duplexing) scheme controlled by the master. The master sends packets to slaves in even-numbered slots triggering a transmission from slaves in the subsequent slot. Slaves are allowed to send packets only in response to a master packet. The way in which the master schedules packets transmission to slaves or polls them determines system performance. We consider the problem of designing an efficient and simple polling and scheduling scheme for Bluetooth. We propose some practical schemes and compare their performance with some ideal schemes derived from known results for polling systems.

The standard end-to-end flow control implemented by the TCP protocol is ill-suited when it comes to achieving fair bandwidth allocation among competing TCP flows. Indeed, the lack of feedback from intermediate nodes does not allow a TCP source to regulate its throughput so that it is not sending more than its fair share, thus penalizing other, less aggressive flows. We propose a novel scheme that, building on existing work on network-layer stateless fair queueing, extends the approach to the TCP layer. Also, we discuss a possible implementation of both the network-layer and the transport-layer architecture. We test our solution under different traffic scenarios and show that not only is a fair bandwidth allocation achieved, but the overall network utilization for TCP flows is also increased.

Target detection in foliage-penetrating (FOPEN), ultrawideband synthetic aperture radar (UWB SAR) images is a challenging problem. Given the low signal-to-clutter ratio, conventional detection algorithms perform poorly in FOPEN SAR images. Using symmetric alpha-stable (S/spl alpha/S) densities to model the impulsive noise, we have developed a region-adaptive automatic target detection (ATD) algorithm. The image is first segmented, and the resulting labeled image is exploited by a region-adaptive target detection algorithm. We evaluate the performance of the algorithm in different frequency bands, and determine which subbands are useful for image segmentation and target detection.