Aik Chindapol

Bit-interleaved coded modulation with iterative decoding and 8 PSK signaling

We have suggested bit-interleaved coded modulation with soft decision iterative decoding (BICM-ID) for bandwidth-efficient transmission over Gaussian and fading channels. Unlike trellis coded modulation, BICM-ID has a small free Euclidean distance but large diversity order due to bit interleaving. With iterative decoding, soft bit decisions can be employed to significantly improve the conditional intersignal Euclidean distance. This leads to a large coding gain, comparable to that of turbo TCM, over both Gaussian and Rayleigh fading channels with much less system complexity. We address critical design issues to enhance the decoding performance and provide the analytical bounds on the performance with an ideal feedback assumption. We investigate the performance characteristics of BICM-ID through extensive simulations and show that at high signal to noise ratios, the performance of BICM-ID converges to the performance assuming error-free feedback.

A new multimedia packet loss classification algorithm for congestion control over wired/wireless channels

In a wireless network environment, common channel errors, due to multipath fading, shadowing and attenuation, may cause bit errors and packet loss quite different from the packet loss caused by network congestion. In congestion control, the packet loss information can serve as an index of network congestion for effective rate adjustment; therefore, wireless packet loss can mistakenly lead to dramatic performance degradation. The paper proposes a packet loss classification algorithm based on detecting the trend in relative one-way trip time (ROTT) when it falls in the ambiguous zone where the packet loss classification is not straightforward. We show that the proposed algorithm greatly benefits rate-based congestion control algorithms for multimedia over IP networks.

An Embedded Packet Train and Adaptive FEC Scheme for VoIP Over Wired/Wireless Ip Networks

Voice over IP (VoIP) has become the fastest growing wireless alternative to conventional telephony service by way of ongoing deployment of WLAN hotspots and even powerful WiMAX coverage. Resulting from the wired/wireless combined best-effort based heterogeneous IP networks which provide more fluctuation in available bandwidth and end-to-end delay, the performance of VoIP quality, especially using the handheld wireless devices, has been greatly degraded due to frequent packet loss and longer delays. This paper proposes a real-time embedded packet train probing scheme for estimating end-to-end available bandwidth so as to accomplish effective congestion control. By trading acceptable delays with adaptive packetization of voice bitstreams, as well as adaptive insertion of forward error correction (FEC) packets, an optimized system driven QoS approach for VoIP can thus be achieved

Performance analysis of coded modulation with generalized selection combining in Rayleigh fading

We consider coded modulation with generalized selection combining (GSC) for bandwidth-efficient-coded modulation over Rayleigh fading channels. Our results show that reception diversity with generalized selection combining can conveniently trade off system complexity versus performance. We provide a number of new results by calculating the cutoff rate, and by deriving analytical upper bounds on symbol-interleaved trellis-coded modulation (TCM) and bit-interleaved-coded modulation (BICM) with GSC. All are verified by simulation. We show that our new bounds on TCM with GSC, which includes maximum ratio combining and selection combining as special cases, are tighter than the previously derived bounds. A new asymptotic analysis on the pairwise error probability, which can be used as a guideline for designing coded modulation over GSC channels, is also given. Finally, we show that BICM with iterative decoding (BICM-ID) can achieve significant coding gain over conventional coded modulation in a multiple-receiving-antenna channel.

Link Layer Packet Loss Classification for Link Adaptation in WLAN

Voice over IP (VoIP) and high quality video streaming becomes an alternative to conventional telephony and TV service in many locations by way of ongoing deployment of WLAN hotspots. The newly proved IEEE 802.11e QoS enhancement standard will be a proper solution of delay demanding for real-time multimedia applications, while the existing auto rate fallback (ARF) link adaptation scheme, designed to improve wireless transmission by dynamically selecting transmission rates and frame sizes, cannot effectively solve the QoS concerns of multimedia communication due to the growing UDP traffic. In this paper, we presented comprehensive analyses of link layer behavior to infer the cause of packet loss in a WLAN. More specifically, we classify the packet loss into either congestion or wireless errors based mainly on delays and MAC layer parameters under congestion and/or link error dominated status. A link layer packet loss classification scheme can thus be proposed to support link adaptation for VoIP applications in WLAN.

Adaptive FEC Scheme For Layered Multimedia Streaming over Wired/Wireless Channels

In wireless communication, noise and channel fluctuation often cause bit errors and subsequently the packet loss, which has different characteristic from the loss from network congestion. For most congestion control algorithms, the packet loss information serves as an index of network congestion and is used for effective rate adjustment; therefore wireless packet loss can mistakenly lead to dramatic performance degradation. We discuss the reasons leading to packet loss and propose a packet loss classification (PLC) algorithm that is based on trend detection of relative one-way trip time. With the assistance of PLC, not only can the wireless loss be separated from the congestion loss, the wireless packet error rate can be also estimated. With the combined information, the new adaptive congestion control algorithm is constructed so that a receiver can acquire an appropriate share of bandwidth. The transmitted data is protected by adaptive maximum distance separable erasure codes according to the wireless channel condition and end-to-end available bandwidth

Distributed Dynamic Resource Allocation for OFDMA Transmissions with Reuse Partitioning

In this paper, we consider the dynamic resource allocation problem for orthogonal frequency division multiple access being used in modern broadband wireless systems such as IEEE 802.16 (WiMax). The objective is to allocate the network resources in an efficient manner in order to maximize the system throughput while guaranteeing the minimum level of quality-of- service to all users under practical conditions such as intercell interference. In order to reduce the communication overhead and computational complexity, we propose a suboptimal two-step approach. Firstly, the radio network controller solves the large scaled network planning problem based on aggregated information about all users in the network. Given the network planning results, each base station solves the small scaled cell throughput maximization problem based on its full knowledge of all users within its cell. We demonstrate the improvements of our scheme using the IEEE 802.16 (WiMax) system as an example. We show that with our proposed dynamic allocation scheme, it is possible to achieve the significant throughput gain with less complexity and signaling overhead.