Rajesh S. Pazhyannur

WLAN-GPRS integration for next-generation mobile data networks

The ongoing wireless LAN standardization and R&D activities worldwide, which target bit rates higher than 100 Mb/s, combined with the successful deployment of WLANs in numerous hotspots justify the fact that WLAN technology will play a key role in wireless data transmission. Cellular network operators have recognized this fact, and strive to exploit WLAN technology and integrate this technology into their cellular data networks. For this reason, there is currently a strong need for interworking mechanisms between WLANs and cellular data networks. We focus on these interworking mechanisms, which effectively combine WLANs and cellular data networks into integrated wireless data environments capable of ubiquitous data services and very high data rates in hotspot locations. We discuss the general aspects of integrated WLANs and cellular data networks, and we examine the generic interworking architectures that have been proposed in the technical literature. In addition, we review the current standardization activities in the area of WLAN-cellular data network integration. Moreover, we propose and explain two different interworking architectures, which feature different coupling mechanisms. Finally, we compare the proposed interworking architectures, and discuss their advantages and drawbacks.

Class and channel condition-based scheduler for EDGE/GPRS

The efficient management of the radio resource of a 3-G system is important from an operators perspective. This, however, cannot be the only concern when quality of service (QoS) negotiations have been made for various users and the operator has to uphold these. This leads to a fairness objective that the operator has to keep in mind. In this paper we outline a scheme to perform packet-level scheduling and resource allocation at the wireless node that takes into account the notions of both efficiency and fairness and presents a means to explore the trade-off between these two notions. As a part of this scheme we see the scheduling problem as deciding not just the packet transmission schedule but also the power allocation, the modulation and coding scheme allocation and the spreading code determination since the latter three directly influence the radio resources consumed. Using a utility maximization formulation based on the data-rates that the mobiles can transmit at, we decide on the weights for a weighted proportionally fair allocation based scheduling algorithm. We also show how one can adapt the weights and the algorithm for a time-varying channel. We conclude with a simulation based performance analysis for infinitely-backlogged sources and TCP sources on an EDGE system.

PPPmux-a new protocol for transporting small IP packets

We introduce a new protocol to improve the capacity of communication links to carry real-time applications that generate a large number of small packets. The key issue in the transport of such applications is the relatively large protocol overhead for each packet. For example, the overhead for RTP/UDP/IP/PPP/HDLC is 12/8/20/7 bytes respectively resulting in a total of 47 bytes. This is considerable for applications with a small payload (10-20 bytes). RTP/UDP/IP header compression (cRTP) mitigates the problem partially by reducing the corresponding protocol overhead from 40 bytes to a best case of 2 bytes. PPPmux provides a further improvement by multiplexing multiple application-level packets into a single PPP frame thereby reducing the PPP overhead per packet. One potential deployment of PPPmux is in cellular networks (IS-95A and UMTS) on links connecting base stations to the radio access network controllers (RNC). The paper provides numerical results listing capacity improvements over existing schemes. Since, multiplexing concatenates multiple IP packets into a single PPP frame, a frame loss would result in loss of multiple IP packets. We provide guidelines on the choosing the optimal multiplexing size that gives the best trade-off between increase in link capacity and increase in packet loss. Specifically, the multiplexing size of 200 bytes gives the best compromise between capacity and packet loss for typical cellular applications.

Asymptotic results for voice delay in packet networks

This paper investigates the performance of packet networks carrying many low bit-rate real-time sources of traffic. We are primarily motivated by applications to the design of wide-area transmission links carrying coded voice traffic between the base sites and the core network of a cellular system. The large number of such links and distances they span lead to significant recurring costs for cellular operators forcing them to operate these links at very high utilization. The system model is composed of a highly utilized transmission link fed by a large number of voice sources. The voice sources are characterized by their correlation structure and relative time-offset with respect to each other. We present simulation and analytical results for this system emphasizing the impact of system utilization, correlation structure, etc. The analysis is based on two different approaches (1) Gaussian queue analysis and (2) heavy traffic analysis. The analysis reflects the statistical properties of the voice sources, in particular, their correlation structure. A key analytical result is that the delays in the system can be characterized by two scaled exponential distributions: one for small delays and the other for large delays.

Two-law approximation for voice delay in packet networks

Traditionally, voice has been transported using circuit switched networks, the Public Switched Telephone Network (PSTN) for example. However, driven by the ubiquity of the Internet and the development of low bit-rate digital voice codecs there has been increasing focus on using packet-switched networks for voice traffic. We focus on one such application. Our model comprises of a relatively slow packet link (between 1.5 and 5.0 Mbps) being utilized for voice traffic. Large number (of the order of 100) voice sources are multiplexed on this link. For such a link we obtain delay distributions seen by a voice source. Specifically, we exploit key characteristics of the model, such as the large number of sources to obtain a heavy traffic approximation for the system. The key result is that the delays can be well approximated by concatenation of two exponential distributions. We also provide valuable insight into how the delay distribution is connected with the statistical properties of the voice sources, in particular their correlation behavior. Our analytical results are validated with simulation results.