Network technologies and air Interface

Project goal

Comparison of 3GPP Release 10 and 802.16m characteristics in Opnet Modeler.  Implementation of MU-MIMO and scheduling algorithms for MU-MIMO for 3GPP Release 10 and Wimax in Opnet Modeler.

This project is supported by grant of Nokia Siemens Networks in Russia to Mikhail Gerasimenko for studies abroad (at Tampere University of Technology).

Introduction

NS-2 Network simulator is being used in huge amount of scientific research. Nokia Siemens Networks uses customized clone of this simulator. This clone is able to produce output about bursts as SVG picture series.  The main goal of this project is to develop a tool for putting SVG images from series together and playing it as a video stream with forward/back navigation. We need to consider irregular time intervals between SVG images at the stage of playing the series of images as a video stream.

Introduction

The main goal of this project is to investigate abilities for seamless WLAN off-loading of 3GPP networks. It implies theoretical and practical investigation, such as abstract mathematical model, simulations, field testing and so on. We need to figure out how these networks could be used simultaneously and which characteristics they have.

Introduction

Based on the IEEE 802.11 network equipment there might be designed comparatively cheap systems for video transmissions by wireless. However, if any video sources are available in a such system, there might be the problems which cause an inefficient utilization of the channel. It is complicated to determine what is the actual number of the sources of the video can be used in the system using only the parameters of a manufacturer of the network equipment and video codecs properties. The purpose of this research is to evaluate efficiency of the channel utilization and to propose the methods how to improve it.

Environment

We consider the system of the multiple access to shared wireless communication channel , which includes a large number of similar type users (digital video cameras) and the receiving/processing center. The connection of an abonent with the center is supported by the wireless channel based on the IEEE 802.11 standard. In this case the users transmit the precompressed data via the WiFi adapters and receiving/processing center receives this data through access point. We make the following assumptions:

The capacity (throughput) of the channel is sufficient for the overall stream transmission from M abonents provided that we do not include the overhead caused by the multiple access problems (channel utilization by several abonents).

The video stream from each abonent camera can be viewed as a sequence of the comparatively short inter frames which are mixed with the longer key (intra) frames (for video transmission the MPEG2, MPEG4 and H.264/AVC are used)

Research directions

Estimation of actual system throughput.

The basic mode which is supported by all the 802.11 standard equipment is Distributed Coordination Function (DCF) mode. This mode is based on the random multiple access to the channel. In a DCF access some conflicts may arise. They cause the situation when the actual number of the users in the system should be reduced to fulfill the mentioned in assumption. Some new methods should be worked out in order to estimate the allowed number of users in this system with conflicts. These methods should take into account:
the set of requirements to the video quality (QoS) received by receiving/processing center;
statistical properties of the compressed video stream according to the characteristics of the used bit-rate control algorithm;
characteristics of the collision detection algorithm implemented in IEEE 802.11 equipment.

Introduction of methods to improve the actual throughput of the system.

The goal of this research is to achieve either the best possible quality of the transmitting video or the maximum number of the users in the system for a limited channel resources to upgrade the basic system.

If case of no motion the camera transmits rather small inter frames. During such transmission the collisions are seldom. If there is motion then the size of frames increase. If motion is intensive then the key frames is formed. The size of a key frame is 4..8 times bigger than the size of an inter frame. If several cameras form the key frame simultaneously and transmit it, then in most cases the collision occurs. This collision will be resolved for a long time.

There are three ways to avoid such collision and to improve the system throughput. The first way is to use desynchronization scheme for transmitting the key frames to several users.

The second way is to redistribute the channel resources considering the users with intensive motion in the camera input by using video compression rate control algorithm.

The third way is to avoid using the random access mode and to use the mode with access point coordinating the users network - Point Coordination Function (PCF) mode. In this mode collisions are completely eliminated, but a part of the channel resources will be spent for transmitting the control frames from access point to the users.

Project stages

In the process of working out the project the theoretical research should be executed, the demo stand should be created in order to demonstrate the proposed approach. It should be taken into consideration that most network adapters and access points do not support the Point Coordination Function mode. The corresponding equipment should be provided to create such a stand for demonstration purposes.

Project team

Victor Malishev, professor
Boris Lavrenko, postgraduate student
Anastasiya Tsyba, postgraduate student
Saint-Petersburg State Electrotechnical University

Introduction

Today wireless technologies are increasingly applied to transmit a human speech. As number of wireless users seeks to grow, improved trade-off between network capacity and high speech quality is in greater demand than ever before.

One of the most important factors of effective exploitation of channel capacity is choice of the most proper algorithm of coding/decoding speech information – codec. Codec should represent an audio signal with a minimum bit rate while maintaining perceptual quality. There is always a trade-off between the bit rate and the quality. It’s need to note that usually codecs developed for packet switched network are designed for wired technologies. So it’s became the subject of consideration how such codecs will be suitable for wireless network with random delays of packets and bit errors.

Currently, a great number of various speech coding schemes are exist. All of codecs can be broadly divided into 3 groups: waveform codecs, source codecs and hybrid codecs. Typically waveform codecs are used at high bit rates, and give very good quality speech. Source codecs operate at very low bit rates, but tend to produce speech which sounds synthetic, such codecs use harmonic synthesis of signal based on information about it’s vocal components – phoneme. Hybrid codecs use techniques from both source and waveform coding, and give good quality speech at intermediate bit rates. And the trade-off between bandwidth and quality in hybrid codecs is much better. That’s why this group of codes is of interest.

The hybrid codecs have different coding algorithms, so they have different complexity of algorithm, packet lengths, the delay introduced, speech quality measure. Also they differently respond to channel errors or packets loss.

Another classification of hybrid speech codecs accomplishes in operating bit rate. Some codecs have a fixed bit rates but another ones can dynamically respond to varying channel conditions. For example in present research will be consider such codecs as G.729, GSM FR, iLBC with fixed bit rate and EVRC, GSM AMR, Speex with dynamically changed bit rate.

The project concentrated on narrowband speech codecs, designed to provide an efficient digital representation of telephone-like signals, bandlimited to between 200 and 3400 Hz and sampled at 8 kHz, for short range (100…1000 meters) wireless ad-hoc packet switching network based on chirp spread spectrum signals in 2.4 GHz ISM band and 60…80 MHz bandwidth. The throughput of a data link is about 500 kbit per second; packet length varies from 64 to 256 byte. Computational power “on board” is strictly limited.

So this project is aimed to carry out the choice of most suitable speech codec and to fulfill practical implementation of algorithm chosen or may be some modification of selected codec for the wireless network which has specific characteristics such as bit error, random delays and short-term link losses.

Project stages

First of all it’s need to gather all necessary information about speech codecs and to range it. The complexity of algorithm which influences in MIPS of DSP, delay introduced, MOS and other parameters of codecs are of interest. Such theoretical research is considered as initial stage.

Mainly the choice of coding scheme is inseparably connected with channel conditions and parameters of DSP used. So alteration one of this conditions can (but it’s not necessarily) lead to codec change.

Next part of present research deals with practical application of codecs and codec selection. The main interest is to study how different codecs respond to channel error, packet delay and packet loss and here what bandwidth is needed.

To make this study complete it is intended to test the system in real working conditions.

Targets

The main purpose of this project is to determine the most acceptable algorithm of speech coding and to apply them in real network. Such algorithm should be robust with packet loss and channel errors, and should provide the optimum trade-off between bit rate, the quality of reconstructed speech, the complexity of the algorithm and the delay introduced.

Introduction

IEEE 802.16 is a standard for the wireless broadband access network, which provides a flexible fixed-wireless and mobile-wireless access between subscriber stations (SS) and a provider. The main advantages of IEEE 802.16 when compared to other wireless network access technologies are the longer range and a more sophisticated support for the Quality-of-Service (QoS) at the MAC level. Different application and service types can be used in 802.16 networks and the MAC layer is designed to support this convergence.

To request uplink resources, 802.16 stations may send bandwidth request by means of uplink contention. The contention resolution mechanism in the 802.16 networks differs from the one used in the IEEE 802.11 (WiFi) standard. While the WiMAX SS tries to avoid collisions by generating a random backoff timer and listening whether the medium is busy or not, the WiMAX SS just picks up randomly a transmission opportunity. Though the transmission opportunity is chosen based on the similar truncated binary exponential backoff mechanism, an SS does not detect whether another SS is transmitting at the same time or not. If the BS receives successfully the bandwidth request, then it allocates resources for the SS. Otherwise, if a collision occurs or a packet is lost, no action is taken and the SS retransmits the bandwidth request.

In IEEE 802.16 OFDMa PHY, prior to sending the bandwidth request, an SS sends a special CDMA code to request uplink resources for the bandwidth request.

Research directions

The project will concentrate on the resource allocation procedure and contention resolution mechanism details. In particular, the following aspect will be investigated:

1. The CDMA request codes correlation properties;
2. Performance comparison of different receivers for the CDMA request codes;
3. Dependence of the bit error rate and false alarm - missed detection probabilities ratio to the system 
   performance;
4. Design of the new CDMA request codes with improved correlation properties.
5. Partitioning of the CDMA codes into groups for initial ranging, bandwidth requests, and handover

Project stages

The main project development stages will include:

1. Theoretical model of the dependence of CDMA codes to the system performance;
2. Simulating the CDMA request procedure in MATLAB environment;
3. Obtaining simulation results for different codes and partitioning
4. Interface for high-level simulators to utilize the MATLAB results on CDMA code performance
5. Implementing the designed codes in NS-2 802.16 module to study the system performance.

Team members

Alexey Dudkov, University of Turku
Alexander Sayenko, Nokia Siemens Networks

Introduction

Diversity receiving is an effective technique of information recovery when multi way signal propagation and its corruption take place. There are several well studied approaches and schemes of this technique realization. In the proposed project attention focuses at one of them – intermediate frequency (IF) signal combiner. This approach has both advantages and drawbacks. The most attractive features of mentioned approach are low cost, good compatibility with standard equipment and good adaptation ability.
The purpose of the project is signal quality criteria development, effective signal processing algorithms proposition, hardware prototype elaboration and its experimental testing.

Environment

The system considered consists of the next blocks. Two or more signal sources (QPSK, QAM and another modulation formats like OFDM technology can be used), channel emulator set, IF signal combiner, spectrum analyzer. Ideally complete demodulator and BER tester have to be included in system. The middle two blocks should be elaborated by the team. IF combiner itself is supposed to be consists of weighted signal summer, signal phase and amplitude difference discriminators, signal distortion analyzer, voltage controlled endless phase shifter and controller.
Proposed research is destined for point-to-point or point-to-multipoint microwave links or base stations and not for mobile units.

Research Directions

1. Distortions of a signals with typical modulation schemes at multi path propagation.
2. Development of the robust distortion criterions and signal combining weights for a different conditions.
3. Practical realization and checking of signal analysis and in phase weighted combining.

Project Stages

1. Literature analysis. Simulation of multi way propagation and channel model choice. Electronic components analysis. Development of combiner structure. Signals sources emulators elaboration.
2. GMSK, QPSK, QAM including OFDM processed signals phase discrimination technique simulation and elaboration. Choice of practical ways. Signal processing algorithms development.
3. Combiner system simulation and its parameters estimation. Electrical schematics of laboratory prototype elaboration. Physical hardware design and PCB tracing including signal source emulators. PCBs fabrication, mounting.
4. Combiner prototype and signal sources tuning, its parameters measuring at laboratory conditions. Laboratory experiment with complete diversity reception setup. Experimental results summarizing and analysis.

Contacts

Team leader: Vladimir Prikhodko