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Codec-based adaptive QoS control for VoWLAN with differentiated services
Abstract
Voice over Wireless LAN (VoWLAN) is becoming more and more helpful in our life and is expected to be among the most important applications in next generation networks. However, the maximum number of VoIP sessions that a WLAN can ensure is very small. Moreover, when the WLAN reaches its capacity the addition of one VoIP session affects the QoS parameters of all VoIP sessions. In this paper, we propose an adaptive technique to ensure the active VoIP sessions of users with high priority (from a provider perspective). Thus, in order to guarantee the quality of high priority sessions, we propose to downgrade the quality (low but acceptable MOS) of user sessions with low priority by changing their used codecs (e.g., ITU G729 instead of ITU G711). This technique and all related monitoring functions are defined into the proposed session-based QoS management architecture. In order to validate our approach a complete test-bed is made up by which we have performed some feasibility and gain tests.
... Also, VoIP can be deployed in different structures of networks; ad hoc, mesh, and WLAN. Due to the features of VoIP over WLAN (VoWLAN), it is now getting the attention of not only from researchers, but also from Internet Service Providers (ISP) [8]. ...
... Since no specific codec can work well in all network conditions [7], developing codec adaptive techniques have been proposed. Although developing this mechanism is still in its early stage [12], different adapting codec rate schemes were proposed particularly for real-time applications in wired [8], wireless, or WLAN networks. ...
... In [27], a cross-layer adaptive rate control (CLARC) method has been presented for video application. In [8] the proposed framework designed to ensure good quality of service to higher priority users (specified by ISP), and adjust the codec rate of user with lower priority. ...
The next generation technology of Voice over Internet Protocol (VoIP) allows voice to be delivered over the Internet Protocol based networks. The integration of VoIP and Wireless Local Area Network (WLAN) has become popular. However, WLAN architecture was primarily designed to carry data and not for real-time applications, such as VoIP. Hence, there is a need to enhance the quality of VoIP transmission over the WLAN. On the other hand, IEEE 802.11 standard of WLAN provides support to link adaptive (LA) technique. LA leads to multi-rate transmissions, which causes degradation on voice quality. In this paper, cross-layer based adaptive approach is proposed to improve the quality of voice delivery over WLAN by minimizing the multi-rate effects on the quality of service (QoS) of VoIP. The approach in achieving this is by introducing a communication agent between MAC sub-layer and Application layer, by which congestion caused by the multi-rate could be reduced, if not eliminated. Moreover, Encoder adaptation is utilized at the Application layer as codec parameters are adapted according to the change of rate occurs at the MAC sub-layer. Simulation of different scenarios for different codec rates at different transmission rates has been implemented using OPNET. The result of QoS improvements is the evidence for the proposed cross-layer adaptation approach.
... Some mechanisms implement the planning agent in an intermediate node, which can be a media gateway [Galiotos et al. 2002;Trad et al. 2004], a wireless access point performing cross-layer QoS management [Chen et al. 2011;Sfairopoulou et al. 2007;Kawata and Yamada 2006;Tüysüz and Mantar 2010], or a dedicated QoS management node [Gardner et al. 2003;Tebbani and Haddadou 2008]. ...
VoIP calls are sensitive to several impairments, such as delay and packet loss. One way to overcome these
problems is by adaptively adjusting application-layer parameters to keep a minimum speech quality level.
At the heart of self-adaptive systems lies a feedback loop, which consists of four key activities: monitoring,
analysis, planning, and execution. Nevertheless, the existing adaptive approaches to QoS control of VoIP
do not explicitly exhibit this feedback loop. Bringing it to surface can help developers in designing more
robust and human-independent VoIP systems. This survey presents a comprehensive review of the current
state-of-the-art research on speech quality adaptation of VoIP systems at the application layer and some
research challenges on this subject.
... When buffer is out of threshold, it changes the codec rate of a node to prevent possible losses. There are also some other works deal with QoS by changing the codecs of nodes such as in [11]. However, it does not use RTCP feedback to understand the channel quality. ...
The demand for Voice over IP (VoIP) services over wireless local area networks (WLANs) and Quality of Service (QoS) support
in WLANs have grown substantially in recent years. IEEE 802.11e QoS enhancement standard has an important role on this expansion
as well. It provides a suitable solution for delay sensitive real-time multimedia applications. Under the circumstance of
any deterioration of wireless link conditions, IEEE 802.11 PHY/MAC specifications allow mobile nodes to select a proper transmission
rate to optimize bit transmission. However, if a node selects lower transmission rate, packet losses and delays increase due
to multi-rate effect. Therefore, all active calls have a general degradation. In this paper, we propose a fast, efficient,
adaptive, proactive codec and frame size modification algorithm for the optimization of maximum throughput and voice quality
on multi-rate WLANs.
KeywordsVoice over IP-multi-rate WLANs-QoS-IEEE 802.11e
... A similar research undertaking in [7] has shown that VoIP capacity measured in simulations and testbeds can given similar results, provided limitation in the two methods are addressed appropriately. Increasing VoIP capacity on wireless LAN, by changing CODEC, is researched in [8] and [9]. Using buffering to enhance transmission on a mobile IPv6 network has been experimentally evaluated in [10]. ...
VoIP implementations are nowadays the preferred information technology alternative to public switched telephone networks. With dependence on this technology, VoIP quality and performance are critical. In this paper, we implement some commonly used VoIP CODECs on Windows desktop operating systems to evaluate their performance on two versions on IP, namely IPv4 and IPv6. Performance related metrics like throughput, jitter and latency are empirically measured in all combinations of operating systems with various CODECs and IP versions. A test-bed setup is employed to measure performance related metrics. Results obtained show that there is only a slight difference in performance between TCP IPv4 and IPv6 networks, however VoIP throughput, jitter and latency values are significantly different depending on the choice of CODEC, operating system and the IP version.
... In [19], Tebbani and Haddadou try to guarantee the QoS of high priority calls against the low priority ones (where priority is determined by the SLAs agreement that users establish with providers) by "down-switching" codec and packetization interval of one or more low priority calls based on a set of information (call and user ID, call codec, session duration, packet loss and end-to-end delay, etc.). Their focus is not on the multirate effects but on the congestion provoked by the existence of additional VoIP sessions on the cell, and the algorithm focuses on adapting existing low priority calls in favor of the high priority ones. ...
In multi-rate WLANs, users can suffer transmission rate changes due to the Link Adaptation mechanism. This results in a variable capacity channel, which is very hostile for VoIP and can cause serious quality of service (QoS) degradation in all active calls. Various codec adaptation mechanisms have been proposed as a solution to this, as well as to solve congestion problems on WLAN environments. Here, these solutions are presented, categorized according to the adaptation policy and scenario they implement, and evaluated at call-level in terms of the resulting blocking and dropping probabilities, as well as the perceived voice quality. To define a common performance metric, a new index named VGoS-factor is presented, which by combining these capacity and quality indicators can provide an overall view of the capacity vs. quality trade-off of the proposed mechanisms and consequently help in choosing the adequate policy for each scenario.
Codec is an essential component in VoIP technology, and therefore, the dream of achieving perfect call quality in VoIP cannot be fulfilled without addressing the issues related to codecs. In multirate WLANs, users can suffer transmission rate changes due to the link adaptation mechanism. This results in a variable capacity channel, which is unsuitable for hosting VoIP calls and can cause serious Quality of Service (QoS) degradation in all the active calls. Various codec adaptation mechanisms have been proposed as a solution to this problem, as well as to solve the congestion problems on WLAN environments. However, most of them follow a reactive strategy based on packet loss information to switch codecs. In this chapter, a special codec parameter, namely pps, has been analyzed and an algorithm based on proactive strategy is proposed to decrease the pps in the WLAN access points so as to reduce the packet loss due to buffer overflow.
This chapter provides an overview of WLANs and discusses the pros and cons of implementing VoIP applications over these networks. Precisely, it highlights the current design challenges and shortcomings in WLAN that require careful addressing for successful hosting of VoIP calls. This is also substantiated by a detailed description of VoIP-based WLAN test bed for laboratory experiments. A real-time test-bed-based performance analysis is carried out to study some of the most important factors that influence and tune the performance of the VoIP soft phones in WLAN.
As converged services are becoming the standard for corporate networks, concerns have been arising on the best way to ensure efficient use of expensive WAN bandwidth. Achieving excellent quality of multimedia and voice communication requires the allocation of large bandwidth for these services. In most cases, lossy compression codecs are used to reduce bandwidth requirements resulting in lower quality of service. The codec selection is maintained even during low bandwidth utilization. In this paper, we present a codec interface that combines high-fidelity high-bitrate and low-fidelity low-bitrate codecs. Audio packets from the low-bitrate codec are transmitted on the highest priority class of service of differentiated services networks. This ensures that these packets always arrive at destination. Audio packets from the high-bitrate codec are transmitted on the lowest priority class of service. At low bandwidth utilization, the high-fidelity high-bitrate packets will be used and during congestions, these packets would be dropped. This ensures achieving the best possible voice quality during varying network loads. Simulation results measuring the mean opinion score of voice quality show that superior voice quality can be achieved with this method over using standalone codecs while maintaining excellent quality of service for data traffic as well.
VoIP over Wireless LANs is greatly affected by path-loss, RF interference and other sources of signal attenuation in addition to network congestion. The primary factors involved in effective real-time communication, namely delay and loss, must be within certain controlled limits in such a scenario. Wireless LAN Access Points must, therefore, act in tune with the codecs to ensure high quality of the ongoing VoIP sessions. Further deployment of VoIP in opportunistic communication medium like cognitive radio network requires thorough analysis of the factors involved in design and implementation of such networks along with modifications in VoIP parameters. In this paper, the various codec parameters are analyzed with respect to wireless LAN access points. Thereafter, an optimization algorithm is developed based on proactive strategy to keep loss and latency within tolerable limits. The proposed algorithm is hence implemented in the SIP enabled test-bed to provide optimum Voice over IP (VoIP) performance. The proactive strategy is further applied to study VoIP performance over cognitive radio networks. A simulation model is developed for VoIP transmission in cognitive radio networks with two scenarios where the primary and the secondary user involve in VoIP transmission respectively. For each scenario, extensive analysis of both VoIP and CRN parameters is performed and an algorithm is proposed based on codecs and active queues to ensure high throughput of VoIP traffic with enhanced QoS. Results verify significant performance improvement in VoIP call quality under both wireless LAN and cognitive radio network environment. Simulation and measured results have also been included.
IEEE 802.11 wireless local area networks (WLANs) support link adaptation technique (LA) which allows mobile nodes to select a proper transmission rate. Link adaptation algorithms use packet loss rate as an indication of poor channel conditions. However, most of existing link adaptation techniques cannot classify the loss as collision, congestion or channel error loss and they could mistakenly adjust the rate as if the loss was due to error prone channel. In this paper, we present a detailed classification of packet losses and propose our link adaptation scheme for delay-sensitive VoIP applications. Proposed algorithm only uses its local information and can be easily implemented without any changes on the current IEEE 802.11 standard. Simulation results show better perceived voice quality and higher overall throughput than other existing algorithms.
VoIP over Wireless LANs is greatly affected by path-loss, RF interference and other sources of signal attenuation in addition to network congestion. The primary factors involved in effective real-time communication, namely delay and loss, must be within certain controlled limits in such a scenario. Wireless LAN Access Points must, therefore, act in tune with the codecs to ensure high quality of the ongoing VoIP sessions. In this paper, we analyze the various codec parameters with respect to wireless LAN APs. Thereafter, we develop an optimization algorithm based on proactive strategy to keep loss and latency within tolerable limits. Finally, we implement the algorithm in our test-bed to provide optimum Voice over IP (VoIP) performance.
Transmission of voice over packet-based protocol has recently gained momentum. To this effect, Voice over Internet Protocol (VoIP) implementations on current and future versions of the Internet Protocols networks is becoming a convenient and inexpensive technology. This is the technology that is replacing traditional public switched telephone networks (PSTN). However, like any other such technology, VoIP offers both opportunities and has limitations. VoIP requires real time communication, however IP networks, which are at the core of all VoIP infrastructure, is not designed for such real time applications. Delay, latency, compromised data throughput and unacceptable round trip times that are commonly experienced on IP networks all may lead to voice quality degradation. The actual operating system platform on which VoIP application is in operation, network topology and the choice of communication protocols are all critical in ensuring that the VoIP is of acceptable quality. In this paper, five Windows operating systems (desktop and server) are configured with the two versions of Internet Protocol (one at a time) and empirically evaluated for performance difference for VoIP network traffic. The experiments are conducted on a test-bed, and performance related metrics like throughput, round trip time and latency are measured with five VoIP Codec types. The results show that VoIP network performance depends not only on IP version and Codec type, but also on the choice of the operating system.
Voice over Internet Protocol (VoIP) allows one to communicate verbally over a TCP/IP network instead of using traditional telephony methods. It has become an attractive communication mechanism for consumers globally. And with the declining costs in global communications and the brisk uptake of Internet based technology, VoIP usage will gain popularity in the years to come. However, like any new technology, VoIP has its associated positives and negatives. In this paper, some commonly used VoIP CODECs are implemented on a test-bed setup with different operating systems (Windows Vista, 7 and Server 2008) and performance related metrics like throughout, jitter and latency are measured in various scenarios. The results obtained show throughput values vary significantly between some of the CODECs while delay and jitter also vary between operating systems and versions of the Internet Protocol.
In this paper, we propose a QoS control mechanism that estimates the congestion of wireless channels in the wireless LAN and evaluates the voice packet drop ratio based on the estimated congestion status. The mechanism adjusts the voice packet transmission rate according to the evaluated packet loss ratio, and thereby resolves the congestion situation. In this way, the mechanism can keep users' quality of experience levels within acceptable ranges.
Several technical issues make commercial and large voice over wireless local area network (VoWLAN) services difficult to provide. The most challenging issue when voice over Internet Protocol (VoIP) services are ran over IEEE 802.11-based WLANs is the bandwidth inefficiency due to the considerable overhead associated with WLAN packet transmission. In this work, we propose a session-based quality-of-service management architecture (SQoSMA) to overcome the low number of VoIP calls in IEEE 802.11 Wireless LANs and the negative effect of new call addition when the WLAN reaches its capacity. The SQoSMA combines data and control planes to detect VoWLAN QoS degradations and performs either an adaptive audio codec switching or a call stopping to fix VoWLAN issues in a differentiated services manner. In addition, our solution deals with user sessions information, by considering user priority (from its agreement) to guarantee a certain level of its multimedia applications. Performance evaluation using a real test-bed shows that call codec change and call stopping techniques can easily assure high-priority calls with acceptable call blocking probability.
We measured the capacity for VoIP traffic in an 802.11b test-bed and compared it with the theoretical capacity and our simulation results. We identified factors that have been commonly overlooked in past studies but affect experiments and simulations. We found that in many papers the capacity for VoIP traffic has been measured via simulations or experiments without considering those factors, showing different capacity in each paper. After these corrections, simulations and experiments yielded a capacity estimate of 15 calls for 64 kb/s CBR VoIP traffic with 20 ms packetization interval and 34 calls to 36 calls for VBR VoIP traffic with 0.39 activity ratio.
In this paper a scheme is proposed to deal with congestion problems that can arise due to Link Adaptation (LA) in Wireless Local Area Networks (WLANs) in the presence of VoIP traffic. The proposed scheme operates by first determining if LA has resulted in the system becoming congested. If not, no further action is taken. However, if, LA has resulted in an overloaded system, the voice codec of the handset which has undergone LA is adapted so as to restore the system to its earlier state, thereby alleviating the congestion. This is achieved by specifically adapting the codec so as to maintain approximately the same level of medium usage as was used prior to the LA. The proposed scheme was evaluated on an experimental test-bed and results show that the codec adaptation was very effective at overcoming the problem of LA.
Growing demands for the public wireless broadband services will require more capacity than the one provided by IP-based service
providers (ISPs). The increasing popularity of WLANs due to the use of license-free radio spectrum with low-cost, easily deployable,
high-data-rate wireless services, has encouraged service providers to consider their deployment in high density usage areas
such us public hotspots to provide complementary broadband access to their networks and services. In order to provide consistent
QoS control for multimedia applications (VoIP, VoD,...) over hotspots, a Session Initiation Protocol (SIP) based QoS management
architecture is proposed in this article. Performance evaluations are discussed to illustrate the feasibility of the proposed
architecture.
Multirate wireless networks in general share a common,problem for the transmis- sion of VoIP tra‐c, since the rate changes of some of the ∞ows afiect the transmission of all others, which causes an unacceptable voice quality degradation. In this work, an admission control algorithm is combined with a codec selection block to mitigate this negative efiect. This centralized admission control is able to block or drop calls in order to maintain the system stability despite rate changes. Moreover, the integrated codec selection block is able to select the most adequate VoIP codec for each incoming or already active call based on the channel rate used and according to a number,of optimization policies. Several such policies are designed and evaluated in this paper. Results show that this combined adaptive solution provides a beneflcial trade-ofi among,the performances of the difierent codecs in terms of MOS, blocking and dropping probability and cell resource usage. Furthermore, a number of interesting singularities deriving from the multirate nature of the network are identifled. Keywords: VoIP codecs, Mobile Networks, Multiple channel transmission rates, WLANs
In this paper, we contribute on the service-driven management of quality of service and user access in wireless corporate networks. We propose a Service-Level Agreement (SLA) oriented nomadism management architecture with a top-down vision starting by the specification of company objectives and going down to device-level configurations. We define an algorithm for the automatic translation of application-level quality assurance parameters into network-level configuration parameters. A prototype implementation of our solution to the SLA-driven enterprise nomadism management is presented along with preliminary results on the self-adaptive capabilities of our QoS mapping algorithm.
We propose a wireless resource management mechanism with AP-RMS (access point resource management server) that can guarantee minimum available WLAN bandwidth for real-time applications. Since an AP-RMS cooperates with SIP (session initiation protocol) proxy and 802.1x authentication server, it can dynamically manage codec and wireless channel utilization for high priority real-time applications. In the case of congestion at one AP, the AP-RMS forces real-time applications to change into a smaller codec. If the AP-RMS detects that congestion is eliminated, it permits those applications to use a large codec. We evaluate our proposed mechanism to show its effectiveness.
Today, the policy-based management approach is recognized as an efficient solution to simplify the complex task of managing and controlling networks. To this end, the IETF has introduced a reference framework to build PBM systems. However, this framework only addresses the provisioning of relatively long validity period services based on pre-defined service level agreements. Furthermore, very little work addresses the scalability properties of the instantiation of this framework in a real network. This work aims to extend the IETF PBM framework in order to support dynamic provisioning of short term services (end system signaling) as well as an instantiation scheme that is scalable (distributed provisioning of edge routers). This instantiation scheme is based on the distribution of the provisioning process while keeping centralized only the parts that involve critical resources, that is, the bandwidth brokerage. The performance properties of the proposed scheme are then demonstrated throughout both extensive experimentation and a short analytical study. The results of this work are intended to be used as a guideline to help network operators to design scalable PBM system in order to offer to their customers services with QoS assurance in an on-demand basis.
Providing end-to-end quality of service (QoS) support is essential for video delivery over the next-generation wireless Internet. We address several key elements in the end-to-end QoS support, including scalable video representation, network-aware end system, and network QoS provisioning. There are generally two approaches in QoS support: the network-centric and the end-system centric solutions. The fundamental problem in a network-centric solution is how to map QoS criterion at different layers respectively, and optimize total quality across these layers. We first present the general framework of a cross-layer network-centric solution, and then describe the recent advances in network modeling, QoS mapping, and QoS adaptation. The key targets in end-system centric approach are network adaptation and media adaptation. We present a general framework of the end-system centric solution and investigate the recent developments. Specifically, for network adaptation, we review the available bandwidth estimation and efficient video transport protocol; for media adaptation , we describe the advances in error control, power control, and corresponding bit allocation. Finally, we highlight several advanced research directions.
Network Grade of Service Parameters and Target Values for Circuit-Switched Services in Evolving ISDN
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