Providing deterministic packet delays and packet losses in multimedia wireless networks

‘Anytime, anywhere’ communication, information access and processing are much cherished in modern societies because of their ability to bring flexibility, freedom and increased efficiency to individuals and organizations. Wireless communications, by providing ubiquitous and tetherless network connectivity to mobile users, are therefore bound to play a major role in the advancement of our society. Although initial proposals and implementations of wireless communications are generally focused on near‐term voice and electronic messaging applications, it is recognized that future wireless communications will have to evolve towards supporting a wider range of applications, including voice, video, data, images and connections to wired networks. This implies that future wireless networks must provide quality‐of‐service (QoS) guarantees to various multimedia applications in a wireless environment. Typical traffic in multimedia applications can be classified as either Constant‐Bit‐Rate (CBR) traffic or Variable‐Bit‐Rate (VBR) traffic. In particular, scheduling the transmission of VBR multimedia traffic streams in a wireless environment is very challenging and is still an open problem. In general, there are two ways to guarantee the QoS of VBR multimedia streams, either deterministically or statistically. In particular, most connection admission control (CAC) algorithms and medium access control (MAC) protocols that have been proposed for multimedia wireless networks only provide statistical, or soft, QoS guarantees. In this paper, we consider deterministic QoS guarantees in multimedia wireless networks. We propose a method for constructing a packet‐dropping mechanism that is based on a mathematical framework that determines how many packets can be dropped while the required QoS can still be preserved. This is achieved by employing: (1) An accurate traffic characterization of the VBR multimedia traffic streams; (2) A traffic regulator that can provide bounded packet loss and (3) A traffic scheduler that can provide bounded packet delay. The combination of traffic characterization, regulation and scheduling can provide bounded loss and delay deterministically. This is a distinction from traditional deterministic QoS schemes in which a 0% packet loss are always assumed with deterministically bounding the delay. We performed a set of performance evaluation experiments. The results will demonstrate that our proposed QoS guarantee schemes can significantly support more connections than a system, which does not allow any loss, at the same required QoS. Moreover, from our evaluation experiments, we found that the proposed algorithms are able to out‐perform scheduling algorithms adopted in state‐of‐the‐art wireless MAC protocols, for example Mobile Access Scheme Based on Contention and Reservation for ATM (MASCARA) when the worst‐case traffic is being considered. Copyright © 2002 John Wiley & Sons, Ltd.     

Fast code-rate optimization for robust image transmission over lossy packet networks

In this paper, we propose an efficient method for the allocation of Reed-Solomon codes to source symbols, for unequal loss protection. The proposed formulation recasts the multivariate optimization problem into a univariate one, dramatically reducing the computational complexity. Results are shown for image transmission over lossy packet networks, employing the JPEG2000 and SPIHT encoders. The proposed algorithm exhibits performance equivalent to previous methods, while providing a significant complexity reduction.     

Frequency-hop transmission for satellite packet switching and terrestrial packet radio networks

The performance of frequency-hop transmission in a packet communication network is analyzed. Satellite multiple-access broadcast channels for packet switching and terrestrial packet radio networks are the primary examples of the type of network considered. An analysis of the effects of multiple-access interference in frequency-hop radio networks is presented. New measures of "local" performance are defined and evaluated for networks of this type, and new concepts that are important in the design of these networks are introduced. In particular, error probabilities and local throughput are evaluated for a frequency-hop radio network which incorporates the standard slotted and unslotted ALOHA channel-access protocols, asynchronous frequency hopping, and Reed-Solomon error-control coding. The performance of frequency-hop multiple access with error-control coding is compared with the performance of conventional ALOHA random access using narrow-band radios.     

Transporting voice traffic over packet networks

POTS networks are being rapidly superceded by newer, packet-based ones, which allows a greater facility for voice traffic. This article explores the practical issues involved in deploying voice networks over ATM, frame relay and IP. © 1998 by John Wiley & Sons, Ltd.     

Minimum-cost multicast over coded packet networks

We consider the problem of establishing minimum-cost multicast connections over coded packet networks, i.e., packet networks where the contents of outgoing packets are arbitrary, causal functions of the contents of received packets. We consider both wireline and wireless packet networks as well as both static multicast (where membership of the multicast group remains constant for the duration of the connection) and dynamic multicast (where membership of the multicast group changes in time, with nodes joining and leaving the group). For static multicast, we reduce the problem to a polynomial-time solvable optimization problem, and we present decentralized algorithms for solving it. These algorithms, when coupled with existing decentralized schemes for constructing network codes, yield a fully decentralized approach for achieving minimum-cost multicast. By contrast, establishing minimum-cost static multicast connections over routed packet networks is a very difficult problem even using centralized computation, except in the special cases of unicast and broadcast connections. For dynamic multicast, we reduce the problem to a dynamic programming problem and apply the theory of dynamic programming to suggest how it may be solved.     

Simulcast packet transmission in ad hoc networks

In previous work, unequal error-protection techniques have been applied to improve the throughput of a wireless communication system in which a transmission is received by several radios with different capabilities. For instance, these capabilities may correspond to differences in path loss, fading, or interference. By taking advantage of the broadcast nature of the channel, additional messages for the more-capable receivers can be included on transmissions to the less-capable receivers at very little cost (in terms of required energy at the transmitter or error probabilities at the receivers). This technique has been termed simulcasting or multicast signaling. In this paper, we consider the use of these techniques in an ad hoc network. These techniques impact the link throughput, end-to-end throughput, and network connectivity. We investigate how the choice of parameters for the simulcasting technique affects these network performance metrics. The results indicate that a properly chosen simulcasting technique can improve the link and end-to-end throughput in wireless ad hoc networks with only a slight degradation in other metrics, such as network connectivity.     

Energy-efficient packet transmission over a wireless link

The paper considers the problem of minimizing the energy used to transmit packets over a wireless link via lazy schedules that judiciously vary packet transmission times. The problem is motivated by the following observation. With many channel coding schemes, the energy required to transmit a packet can be significantly reduced by lowering transmission power and code rate and therefore transmitting the packet over a longer period of time. However, information is often time-critical or delay-sensitive and transmission times cannot be made arbitrarily long. We therefore consider packet transmission schedules that minimize energy subject to a deadline or a delay constraint. Specifically, we obtain an optimal offline schedule for a node operating under a deadline constraint. An inspection of the form of this schedule naturally leads us to an online schedule which is shown, through simulations, to perform closely to the optimal offline schedule. Taking the deadline to infinity, we provide an exact probabilistic analysis of our offline scheduling algorithm. The results of this analysis enable us to devise a lazy online algorithm that varies transmission times according to backlog. We show that this lazy schedule is significantly more energy-efficient compared to a deterministic (fixed transmission time) schedule that guarantees queue stability for the same range of arrival rates.     

T-MPLS分组传送网分层体系结构的研究

文章首先针对传送业务从基于电路的时分复用(TDM) 业务向基于分组的业务转变对分组传送网的要求,说明了满足这一要求的分组传送网的总体分层结构,然后在此基础上给出了传送多协议标签交换(T-MPLS)分组传送网的分层结构,最后,分析了基于该结构的TDM和IP等业务的综合承载机制.     

On rate control of packet transmission over fading channels

In error controlled packet reception, a packet is received only if its error probability can be kept below a predetermined level. Error probability control is achieved by posing a minimum signal to noise ratio (SNR) threshold with corresponding packet internal coding scheme, which upper-bounds the packet data rate. We first consider packet transmission over a single-user wireless fading channel with additive Gaussian noise. We derive the optimal SNR threshold that maximizes the communication throughput. We show under a set of generous conditions that the optimal SNR threshold in the low-SNR regime is proportional to the transmit power; the ratio depends neither on the packet internal coding scheme nor on the pre-determined error probability level. The result is then extended to packet multicasting where common information is transmitted to a group of receivers over fading channels.     

Proxy-assisted interactive-video services over networks with large delays

While emerging broadband access technologies such as DSL and cable are making multimedia services feasible and economically attractive for end-users, there still exist several hurdles in terms of service sustainability and reliability. Unfortunately, without the desired quality-of-service (QoS) support, tackling these hurdles with traditional solutions is an insuperably difficult task. Yet, novel designs that are proven to be useful in various scenarios may easily fail when the underlying network experiences severe packet loss or delay. Such circumstances are unavoidable in today's best-effort Internet and will likely prevail in the near future as well. A promising approach in satisfying the stringent requirements of delay-intolerant video applications is to benefit from configurable proxies. In this study, we introduce a versatile proxy-based solution to enhance the performance of such applications running over networks with large delays. We first propose a methodology that accurately identifies lost packets in real time. This methodology is then used by the proxy and end-users to improve the error-control/protection capability of the video applications. By Internet experiments between the US and Europe, we demonstrate the effectiveness and potential benefits of the proposed approach.     

TOPS: an architecture for telephony over packet networks

Packet telephony is of increasing interest in both the telecommunications and Internet communities. The emergence of packet telephony will create new services, and presents an opportunity to rethink how conventional telephony services are implemented. In this paper, we present an architecture for telephony over packet networks (TOPS). TOPS allows users to move between terminals or to use mobile terminals while being reachable by the same name. TOPS users can have multiple terminals and control how calls are routed to them. TOPS allows for terminals with a range of capabilities such as support for video, whiteboard, and other media with a variety of coding formats. TOPS retains the necessary information on terminal capabilities to determine the appropriate type of communication to be established with the remote terminal. The architecture assumes that the underlying network supports the establishment of end-to-end connectivity between terminals, with an appropriate quality of service. The components of TOPS are a directory service, an application layer signaling protocol, and a logical channel abstraction for communication between end-systems. The directory service maps a user's name to a set of terminals where the user may be reached. A user can control the translation operation by specifying profiles that customize how his name is mapped to a set of terminals where he can be reached. Terminal capabilities are also stored in the directory service. The application layer signaling protocol establishes and maintains call state between communicating terminals. The logical channel abstraction provides a shared end-to-end context for a call's constituent media and control streams, while isolating the applications from the details of the network transport mechanisms. In addition to supporting simple point-to-point calls, the architecture supports both centralized and decentralized conferencing. We also introduce a simple encapsulation format for voice     

Clock synchronization of MPEG-2 services over packet networks

We address the problem of time-base synchronization for MPEG services in the presence of network jitter. The conventional methods to obtain a stable clock indication are based on Phase Locked Loops (PLLs). They have the disadvantage of a long startup phase and thus are unsuitable for services that require simultaneous accuracy and rapidity. We develop a new time-base synchronization technique based on a Least-square Linear Regression algorithm (LLR). We show that LLR is able to perform time-base synchronization with the same accuracy as of a PLL and with a substantial gain of rapidity. Finally, we discuss the implementation of the LLR technique as an intermediate time-base synchronization level between the network and a generic multimedia application.     

Two dimensional cross-layer optimization for packet transmission over fading channel

In this paper a single-input-single-output wireless data transmission system with adaptive modulation and coding over correlated fading channel is considered, where run-time power adjustment is not available. Higher layer data packets are enqueued into a finite size buffer space before being released into the time-varying wireless channel. Without fixing the physical layer error probability, the objective is to minimize the average joint packet loss rate due to both erroneous transmission and buffer overflow. Two optimization techniques are incorporated to achieve the best solution. The first is policy domain optimization that formulates the data rate adaptation design as classical Markov decision problem. The second is channel domain optimization that appropriately partitions the channel variation based on particular fading environment and carried traffic pattern. The derived policy domain analytical model can precisely map any policy design into various QoS performance metrics with finite buffer setup. We then propose a tractable suboptimization framework to produce different two-dimensional suboptimal solutions with scalable complexity-optimality tradeoff for practical implementations.     

Performance evaluation of packet data services over cellular voice networks

In this paper we develop a Markov chain modeling framework for throughput/delay analysis of data services over cellular voice networks, using the dynamic channel stealing method. Effective approximation techniques are also proposed and verified for simplification of modeling analysis. Our study identifies the average voice call holding time as the dominant factor to affect data delay performance. Especially in heavy load conditions, namely when the number of free voice channels becomes momentarily less, the data users will experience large network access delay in the range of several minutes or longer on average. The study also reveals that the data delay performance deteriorates as the number of voice channels increases at a fixed voice call blocking probability, due to increased voice trunking efficiency. We also examine the data performance improvement by using the priority data access scheme and speech silence detection technique.     

A protocol for adaptive transmission in direct-sequence spread-spectrum packet radio networks

Wireless communication channels may change greatly from one transmission to the next, due to variations in propagation loss and interference. The use of fixed transmission parameters for such channels results in wasted energy when channel conditions are good. Adaptation of the power, code rate, and symbol rate reduces energy consumption and interference caused to other systems. Such adaptation requires information about the characteristics of the channel, which is more difficult to obtain in a packet radio network (PRN) or other mobile ad hoc network than in a typical cellular communication system. We develop methods for providing partial information about the channel state from three statistics that are derived by different subsystems in the receiving terminals of a direct-sequence spread-spectrum PRN. We present and evaluate a protocol that uses this information to adapt the transmission parameters in response to changes in interference and propagation conditions in the network. The performance of the new adaptive-transmission protocol is compared with a system with fixed transmission parameters and with an adaptive protocol that is furnished with perfect knowledge of the channel state at the completion of each transmission.     

Analysis of packet loss in transport layer over ATM networks

In asynchronous transfer mode (ATM) networks, when cells are lost due to congestion, packets containing the lost cells should be retransmitted in the transport layer, which manages the end-to-end communication. The probability that a packet contains at least one lost cell depends on the packet length. It is thus very likely that the performance of the end-to-end communication is influenced by the packet length. In this paper, we analyse packet loss probability and the achievable maximum throughput when a block of data is divided into packets of fixed size and the lost packets are retransmitted based on the selective repeat automatic repeat request (ARQ). Through this analysis, we examine the effect of packet length and peak cell transmission rate on the performance measures mentioned above. © 1997 John Wiley & Sons, Ltd.     

Making transmission schedules immune to topology changes inmulti-hop packet radio networks

Transmissions scheduling is a key design problem in packet radio networks, relevant to TDMA and CDMA systems. A large number of topology-dependent scheduling algorithms are available, in which changes of topology inevitably require recomputation of transmission schedules. The need for constant adaptation of schedules to mobile topologies entails significant, sometime insurmountable, problems. These are the protocol overhead due to schedule recomputation, performance penalty due to suspension of transmissions during schedule reorganization, exchange of control message and new schedule broadcast. Furthermore, if topology changes faster than the rate at which new schedules can be recomputed and distributed, the network can suffer a catastrophic failure. The authors propose a robust scheduling protocol which is unique in providing a topology transparent solution to scheduled access in multi-hop mobile radio networks. The proposed solution adds the main advantages of random access protocols to scheduled access. Similarly to random access it is robust in the presence of mobile nodes. Unlike random access, however, it does not suffer from inherent instability, and performance deterioration due to packet collisions. Unlike current scheduled access protocols, the transmission schedules of the proposed solution are independent of topology changes, and channel access is inherently fair and traffic adaptive     

Minimizing transmission delays in vehicular ad hoc networks by optimized placement of road-side unit

Wireless Networks - In a vehicular ad hoc network (VANET), a road side unit (RSU) is a network traffic transmitter statically placed along the route to facilitate communication between vehicles and...