Wireless IP Networks Research Articles

Robust Packet Video Transport Over Wireless Fading Channels Using a Joint Source-channel Coding Approach

In this paper, we address the problem of robust video transmission over wireless networks. Specifically, we consider packet video transmission over wireless IP networks based on the RTP/UDP/IP protocol stack. Digital video delivered over wireless networks is expected to suffer quality degradation from both packet loss and bit errors in the payload. In this paper, both packet loss and bit errors in the payload are considered and the performance of a joint source-channel coding (JSCC) approach employing forward error-correction (FEC) coding schemes for H.263 + video transmission is studied. Results indicate that with an appropriate JSCC approach, FEC-based error-control techniques can significantly improve the packetization efficiency for a given end-to-end quality requirement and lead to more acceptable video delivery quality over time-varying wireless networks. Another important observation is that with a JSCC approach the fading effects of wireless links upon end-to-end video quality are substantially decreased compared to a system without using channel coding, resulting in attractive robust performance characteristics.

Quality of Service System Approximation in IP Networks

This paper presents Quality of Service analyses in wired and wireless IP networks based on the three popular techniques – RSVP, IntServ, and DiffServ. The analyses are based on a quick approximation schema of the traffic system with static and dynamic changes of the system bounds. We offer a simulation approach where a typical leaky bucket model is ap- proximated with a G/D/1/k traffic system with flexible bounds in waiting time, loss and priority. The approach is applied for two cascaded leaky buckets. The derived traffic system is programmed in C++. The simula- tion model is flexible to the dynamic traffic changes and priorities. Student criterion is applied in the simulation program to prove results. The results of the simulation demonstrate the viability of the proposed solution and its applicability for fast system reconfiguration in dynamic environmental circumstances. The simulated services cover a typical range of types of traffic sources like VoIP, LAN emulation and transaction exchange.

Video quality estimation in wireless IP networks

This article proposes three methods to estimate the distortion deriving from packet losses in wireless video communication. The proposed methods take into account the short-term properties of the encoded video sequences. A suitable set of functions is adopted to model the distortion envelope resulting from multiple losses. The estimated performance is compared with the actual distortion, evaluated by decoding the received sequence with a properly designed decoder. Numerical results confirm the accuracy of the proposed models in approximating the actual Mean Square Error (MSE) for a wide range of loss rates. Some applications of the proposed algorithms are presented.

On the Interplay of Service Proximity and Ubiquity

The IEEE 802.15.4 standard for Low Power Wireless Personal Area Networks (LoWPANs) has emerged as a promising technology to bring the envisioned ubiquitous paradigm, into realization. Considerable efforts are being carried on to integrate LoWPANs with other wired and wireless IP networks, in order to make use of pervasive nature and existing infrastructure associated with IP technologies. Provisioning of service discovery and network selection in such pervasive environments puts heavy communication and processing overhead in networks with highly constrained resources. Localization of communication, through accessing the closest services, increases the total network capacity and increases the network life. We present a hierarchical service discovery architecture based on SSLP, in which we propose directory proxy agents to act as cache service for directory agent, in order to localize the service discovery communication and access the closest services. We also propose algorithms to make sure that service users are connected to the closest proxy agent in order to access the closest service in the vicinity. The results show that our architecture and algorithms help finding the closest services, reduce the traffic overhead for service discovery, decrease the service discovery time, and save nodes' energy considerably in 6LoWPANs.

The performance of TCP/IP over wireless ATM network

This paper investigates the most important packet discarding schemes that have been designed for best-effort traffic in wireless ATM and IP networks, and performs modeling and simulation work based on Early Packet Discarding using a Pair Buffer Allocation schemes on different traffic environments, including a mix of packets sizes, buffers sizes, diverse types of data and traffic flow models with the aim of establishing a framework for enhancing the performance of these schemes to improve the overall throughput of TCP/IP as a transport mechanism over wireless ATM networks. The simulation is carried out using MATLAB Simulink software. The simulation results show that a high effective throughput was obtained which is above 50% as compared with existing scheme which is lesser. The proposed scheme improves the effective throughput performance by reducing cell loss to a minimal level.

The UMTS base station router

This paper reviews the Alcatel-Lucent Universal Mobile Telecommunications System (UMTS) base station router (BSR)—a radio access network (RAN) and core network (CN) in a single network element. While typical cellular UMTS networks are built from a plethora of network elements and are maintained in a hierarchical fashion, the BSR combines all functions of a radio access network and core network in a single network element. Moreover, the BSR can be considered a wireless edge router that bridges between cellular-specific wireless transmissions and IP networks. Integrating RAN functionality in a single network element presents the following technical advantages: improving the responsiveness (i.e., latency reductions), reducing the complexity, and improving the fault tolerance of the system. © 2007 Alcatel-Lucent.

Rate adaptation for wireless video streaming based on error statistics

This paper presents a new rate-control algorithm for live video streaming over wireless IP networks, which is based on selective frame discarding. In the proposed mechanism excess 'P' frames are dropped from the output queue at the sender using a congestion estimate based on packet loss statistics obtained from RTCP feedback and from the Data Link (DL) layer. The performance of the algorithm is evaluated through computer simulation. This paper also presents a characterisation of packet losses owing to transmission errors and congestion, which can help in choosing appropriate strategies to maximise the video quality experienced by the end user.

A Cross-Layer (Layer 2 + 3) Handoff Management Protocol for Next-Generation Wireless Systems

Next-generation wireless systems (NGWS) integrate different wireless networks, each of which is optimized for some specific services and coverage area to provide ubiquitous communications to the mobile users. It is an important and challenging issue to support seamless handoff management in this integrated architecture. The existing handoff management protocols are not sufficient to guarantee handoff support that is transparent to the applications in NGWS. In this work, a cross-layer (layer 2 + 3) handoff management protocol, CHMP, is developed to support seamless intra and intersystem handoff management in NGWS. Cross-layer handoff management protocol uses mobile's speed and handoff signaling delay information to enhance the handoff performance of mobile IP that is proposed to support mobility management in wireless IP networks. First, the handoff performance of mobile IP is analyzed with respect to its sensitivity to the link layer (layer 2) and network layer (layer 3) parameters. Then, a cross-layer handoff management architecture is developed using the insights learnt from the analysis. Based on this architecture, the detailed design of CHMP is carried out. Finally, extensive simulation experiments are carried out to evaluate the performance of CHMP. The theoretical analysis and simulation results show that CHMP significantly enhances the performance of both intra and intersystem handoffs

New approach to designing GPRS location update function

Wireless IP network has attracted significant interest due to their ability to support both voice and data transfer in mobile communication. One of the main issues concerning such network is the analysis and design of mobility function particularly the location management. In this paper, we focus on modelling location update function in wireless network standard GPRS (General Packet Radio Service) by using an agent approach [1].

Fast Handover Failure-Case Analysis in Hierarchical Mobile IPv6 Networks

The fast handover protocol adopted in a IPv6 hierarchical structure provides a seamless handover in wireless IP networks by minimizing the handover latency. To reduce the handover latency, the fast handover uses anticipation based on layer 2 trigger. Nonetheless, a mobile node can still lose its connection with the old link during the fast handover procedures. Accordingly, this paper analyzes the handover latency and packet delivery costs associated with fast handover failure cases based on a timing diagram.

Introducing reconfigurability and cognitive networks concepts in the wireless world

Wireless communications and IP networks will evolve into a new converged vision, which is called beyond the 3rd generation. This vision can be efficiently realized through the advent of cognitive networks. Cognitive wireless networks are capable of reconfiguring their infrastructure, based also on experience, in order to adapt to the continuously changing environment. Main reconfiguration actions, in the PHY/MAC layers, in the selection of the most appropriate technology and spectrum for operation. Such innovative concepts require a holistic combination of a variety of engineering disciplines. This paper addresses such issues. Emphasis is given on business model aspects, the management architecture and functionality for cognitive wireless networks, as well as on the related hardware aspects that enable reconfiguration

Providing fault-tolerant authentication and authorization in wireless mobile IP networks

In wireless Mobile IP systems, the authentications and authorizations are performed by AAA (Authentication, Authorization, and Accounting) servers. An AAA server associates with a mobility agent to form an administrative domain. If an AAA server fails, all the mobile nodes (MNs) within the corresponding domain (failure-effected MNs) are unable to execute data services since their authentications and authorizations cannot be performed by the faulty AAA server. To tolerate the failure of the AAA server, this paper presents an efficient fault-tolerant approach. Once a failure is detected in an AAA server of an administrative domain, the proposed approach utilizes the AAA servers in other administrative domains to virtually generate a backup AAA server. To further reduce the fault-tolerant cost, the proposed approach additionally uses two techniques: preservation and tracking to assist the generation of the backup AAA server. Due to introducing these two techniques, the proposed approach does not need to retrieve the AAA records of failure-effected MNs while performing fault tolerance. Finally, we use M/ G/ c/ c queuing model to analyze the effectiveness of the proposed approach over previous approaches. The analytical results are also validated by simulations.

QoS Provisioning in Wireless IP Networks

This paper deals with quality of service (QoS) provision in wireless IP networks. QoS provision is particularly challenging in wireless networks, where network resources are generally limited, variable over time and shared. In the design of possible measures to assure QoS one should consider that standardization is well established for the network layer Internet Protocol and for many underlying technologies of frequent use (e.g. IEEE 802.11, BLUETOOTH or HIPERLAN II). Therefore, as far as research on QoS is concerned, there is little room in both the IP and the link-layers for improved IP over wireless interfaces. In this paper we illustrate a solution in which an intermediate Wireless Adaptation Layer (WAL) is transparently interposed between the IP layer and specific link-layer technologies as a solution to provide QoS. The WAL addresses two main issues: (i) compensation for channel impairments in different platforms in order to enhance wireless channel reliability and (ii) implementation of traffic control and packet scheduling mechanisms to satisfy bandwidth and delay requirements, as well as to enforce a general principle of fairness among the IP associations contending for network resources and achieve optimal exploitation of transmission capacity. The WAL consists of a set of modules, each one in charge of a specific task, which can be enabled or disabled depending on the specific network environment. The novelty of the WAL approach is its capability of adapting itself to different wireless interfaces selecting performance enhancing modules for specific networks. This requires to modify the standard TCP/IP protocol stack by introducing an intermediate layer between the IP layer and the Data Link layer, with performance enhancement purposes. This paper focuses on two modules in particular, namely a traffic control module, which is in charge of performing congestion control and channel state dependent scheduling (CSD) packet scheduling, and a forward error correction (FEC) module, which compensates for channel impairments. This paper presents the proposed architecture provided with these modules and reports some measurements and simulations highlighting benefits resulting from the use of such modules.

Traffic Analysis and Video Quality Evaluation of Multiple Description Coded Video Services for Fourth Generation Wireless IP Networks

For the performance evaluation of future wireless communication systems, such as the fourth generation wireless networks, traffic traces of realistic services are needed. Multiple description coding (MDC) is gaining a lot of interest lately and is a viable solution to provide robust video services over single or multi hop wireless networks and MDC introduces more flexibility for network coding. Furthermore it has the capability to support heterogeneous terminals as they are accepted to be used in 4G wireless networks. By means of twelve well known video sequences, in different video formats, we generate the frame size traces and evaluate the multiple description coding characteristics. In addition to that we highlight the expected overhead due to the underlying RTP/UDP/IP protocol suite. As an objective quality measurement at the application layer, we investigate the video quality in dependency of lost and error-prone descriptors. This allows researcher to convert the network losses of their network models directly into video quality values. This step makes the work unique as single layer coded video streams would always need further postprocessing to retrieve the video quality.

Performance analysis of double buffer technique (DBT) model for mobility support in wireless IP networks

Existing mobility support models in cellular communications misinterpret mobility loss in cellular networks as congestion loss, thus it degrades the performance by invoking unnecessary congestion control action. In this paper, we investigated the performance of Double Buffer Technique (DBT) model for mobility support in wireless IP networks. The DBT model uses the END message and the TQRS timer to maintain the packet sequence and decrease the load on the new foreign agent when the timer expires, respectively. Also, the protocol showed improved performance degradation caused by the handover of the mobile terminal. In order to demonstrate the superiority of our scheme over the existing ones, we used the following performance metrics: packet out-of-sequence, cell loss ratio, bandwidth overhead, and suitability for real-time services. The numerical results obtained revealed that the buffer size, the waiting time, and the packet loss probabilities in the model were suitable to the wireless IP environment.

Joint loss pattern characterization and unequal interleaved FEC protection for robust H.264 video distribution over wireless LAN

The recently adopted H.264 standard achieves efficient video encoding and bandwidth savings. Thus, designing communication protocols and QoS control mechanisms for H.264 video distribution over wireless IP networks is a topic of intense research interest. Delivering video streams to terminals via a wireless last hop is indeed a challenging task due to the varying nature of the wireless link. While a common approach suggests exploiting the variations of the wireless channel, an alternative is to exploit characteristics of the video stream to improve the transmission. In this paper, we combine both approaches through an efficient wireless loss characterization and a low-delay unequal interleaved FEC protection. Besides deriving new QoS metrics for FEC block allocation, the wireless loss characterization is as well used to adjust the interleaving level depending on the loss correlation exhibited by the wireless channel. This novel unequal interleaved FEC (UI-FEC) protocol allows graceful video quality degradation over error-prone wireless links while minimizing the overall bandwidth consumption and the end-to-end latency.

QoS Provisioning with New Effective Bandwidth/Buffer Calculation Scheme in Wireless IP Networks

The huge commercial success of mobile telephony, the phenomenal growth of Internet users, the popularity of IP-based multimedia applications are the major driving forces behind third-generation (3G), ongoing Byond 3G (B3G), and forth-genertion (4G) evolution. 3G brought wired applications, both data and multimedia, into wireless environments. It operates on IP-based infrastructures to provide wider service access capability. To support and satisfy QoS (Quality of Service) of diverse IP-based multimedia applications, traffic management, such as Connection Admission Control (CAC) and resource allocation, becomes essential. CAC and resource allocation are computationally complex when combined with QoS guarantee for traffic with different characteristics. However, CAC and resource allocation are real-time traffic control procedures. Hence, processing load should be minimized to reduce delay. At the same time, network resources should be utilized efficiently to accommodate more users. However, reducing processing load and obtaining high resource utilization efficiency has been considered to be contradictory matter. In addition, CAC and resource allocation schemes which consider multiple QoS criteria --- loss and delay --- simultaneously have not been adequately studied. Simultaneous QoS consideration is important to satisfy stringent and diverse QoS requirements of multimedia traffic. In this paper, we propose a nobel effective bandwidth/buffer calculation method based on a virtual channel/buffer analysis scheme. We show that our method can achieve high resource utilization efficiency with reduced processing load. Moreover, we show that our scheme allows for simultaneous consideration of multiple QoS criteria, loss and delay.

A Header Compression Algorithm for Real-Time Streams in Wireless IP Networks

In wireless IP networks, it is necessary to adopt header compression schemes for the reduction of the protocol header size in order to save the scarce wireless bandwidth resource and make real-time services economically feasible and physically realizable. However, all the existing header compression schemes have not taken wireless channel state into consideration when designed. For a better usage over wireless links, a new channel state based robust header compression algorithm (CSB-ROHC) for real-time streams in wireless IP networks such as 3G platforms is proposed and analyzed in this paper. Through adjusting the dimension of VSW in W-LSB encoding (a key parameter in header compressor) with the accurate estimation of wireless channel state, this new algorithm can achieve a good balance of compression ratio and error-resistant robustness for its adaptive usage over wireless links. The authors present simulation results that demonstrate the effectiveness of the CSB-ROHC algorithm over wireless links.

Performance Analysis of Double Buffer Technique (DBT) Model for Mobility Support in Wireless IP Networks

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REDUP: a packet loss recovery scheme for real-time audio streaming over wireless IP networks

Due to the characteristics of (1) smaller bandwidth and (2) unreliable transmission media, real-time audio streaming over wireless networks is not trivial. To have smooth audio streaming over wireless networks, we propose a scheme called REDUP in this paper. Two sending modes, in which redundant information is embedded in each packet, that the REDUP scheme contains are (1) the ''redundant'' mode and (2) the ''duplicated'' mode. Let a packet i can contain three audio frames i, i-1, and i-2. In the redundant mode, frame i uses a codec of better quality than that for frames i-1 and i-2. In the duplicated mode, frames i, i-1, and i-2 use the same codec, which has lower quality than that for frame i used in the redundant mode. The ''redundant'' mode may give better quality of sound but consumes more bandwidth, while the ''duplicated'' mode gives lower quality of sound but consumes less bandwidth. The two modes are selected depending on the networking situation. Round trip time (RTT) between the mobile host and the mobile gateway is used to determine the networking situation. Two thresholds named ''upper-ratio'' and ''lower-ratio'' are set. When the average RTT exceeds the upper-ratio multiplies the maximum RTT, the networking situation is set to congested; when the average RTT is under the lower-ratio multiplies the maximum RTT, the networking situation is set to unloaded. The transmission mode is switched from ''redundant'' to ''duplicated'' when the networking situation is determined to be congested. The transmission mode is switched from ''duplicated'' to ''redundant'' when the networking situation is determined to be unloaded. In this way, the redundant information of packets is adapted to the networking situation and the bandwidth can be utilized more effectively.