Codec-based adaptive QoS control for VoWLAN with differentiated services

Tebbani, Badis; Haddadou, Kamel

doi:10.1109/wd.2008.4812886

Cited by 14 publications

(13 citation statements)

References 8 publications

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“…Therefore, we defined the capacity for VoIP as the maximum number of wireless nodes so that both uplink and downlink delay do not exceed 60 ms. Moreover, as shown in our previous works [4,23], the uplink delay added less than 1 ms of transmission delay, which allows us to assume that the end-to-end delay is essentially the same as the downlink delay. We also defined the capacity for VoIP as the maximum number of wireless nodes so that the jitter delay did not exceed 10 ms, and therefore, the MOS did not fall below 3.6.…”

Section: Performance Evaluation Of Vowlan Servicesmentioning

confidence: 86%

“…One of the most problematic unsolved issue concerning VoWLANs is bandwidth inef f iciency [1][2][3]. For instance, ten calls with the G.729 codec can hardly be supported on 802.11b with acceptable quality of service (QoS), even without competing traffic-less than 2% efficiency [4]. Furthermore, when the WLAN reaches its capacity, the addition of one VoIP call does not only affect its own QoS parameters (delay, packet loss, .…”

Section: Introductionmentioning

confidence: 99%

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Quality assurance of voice over WLANs (VoWLANs) with differentiated services

2010

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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 inef f iciency 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.

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Section: Performance Evaluation Of Vowlan Servicesmentioning

confidence: 86%

Section: Introductionmentioning

confidence: 99%

Quality assurance of voice over WLANs (VoWLANs) with differentiated services

2010

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“…Tebbani and Haddadou [12] have investigated congestion of voice communication over wireless LANs and have proposed an adaptive technique of selecting compressed codecs for lower priority calls. This adaptive technique factors in user priorities and current wireless LAN congestion.…”

Section: B Related Workmentioning

confidence: 99%

“…The proposed schemes in [12]- [15] are focused on resolving the types of errors and packet losses exhibited by wireless networks and do not consider the services that can be provided by WANs and MPLS networks.…”

Section: B Related Workmentioning

confidence: 99%

Quality codec interface for VoIP on differentiated services networks

Zmily

Abu-Saymeh

2013

2013 International Conference on Smart Communications in Network Technologies (SaCoNeT)

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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.

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“…A similar research undertaking in[7] has shown that VolP capacity measured in simulations and test beds can given similar results, provided limitation in the two methods are addressed appropriately. Increasing VolP capacity on wireless LAN, by changing CODEC, is researched in[8] and[9]. Using buffering to enhance…”

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confidence: 99%

VoIP network performance evaluation of operating systems with IPv4 and IPv6 network implementations

Narayan

Gordon

Branks

et al. 2010

2010 3rd International Conference on Computer Science and Information Technology

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Ahstract-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. Windows operating system, CODEC, operating systems I. I NTRODUCTION Information Technology continue to rigorously enhance all aspects of personal, business and social activities in the twenty-first century living. Times are changing, technology is changing, and so do the demands and the expectations of millions of global users. Undoubtedly, the Internet is playing a vital role by providing a global information superhighway unrivalled by any other technological advancement. Internet Protocol (IP) version 4 currently forms the Internet cloud and has numerous limitations -this eventually will be replaced with version 6. The new version addresses inherent problems in the earlier version, and it offers new opportunities that can enhance communication experiences of users beyond current scope. However, IPv6 has larger data overhead than its counterpart, which hints network performance issues. In this paper, we performance analyze two commonly used Windows desktop operating systems, implemented with different versions of Internet Protocol, for three different voice CODEC types. The empirical experiments are conducted on a test-bed, and performance related metrics like throughput, delay, latency and jitter are measured. Our 978-1-4244-5540-9/10/$26.00 ©2010 IEEE 669 results show voice CODEC types give different performance metrics values depending on operating system and IP version.The rest of the paper is organized as follows: Section II describes Voice over IP CODECs, similar work undertaken by other researchers is mentioned in Section III, and Section IV outlines the experimental setup used in this research. We present the results and discuss the findings in Section V.Finally, the research is concluded in Section VI. II. B ACKGROUNDVoIP, also commonly known by descriptors such as IP telephony and broadband phone uses TCPIIP networks to establish voice communication between devices. Since voice is analogue and the TCP/IP media is digital, CODEC is used to compress analog voice signal to digitally encoded version that can travel on computer networks. Sound quality, bandwidth required, and resource requirements all depends on the choice of CODEC. G.7xx, inclu...

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Codec-based adaptive QoS control for VoWLAN with differentiated services

Cited by 14 publications

References 8 publications

Quality assurance of voice over WLANs (VoWLANs) with differentiated services

Quality assurance of voice over WLANs (VoWLANs) with differentiated services

Quality codec interface for VoIP on differentiated services networks

VoIP network performance evaluation of operating systems with IPv4 and IPv6 network implementations

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