COMPARATIVE STUDY FOR PERFORMANCE ANALYSIS OF VOIP CODECS OVER WLAN IN NONMOB...Zac Darcy
Voice over IP (VoIP) applications such as Skype, Google Talk, and FaceTime are promising technologies for providing cheaper voice calls to end users over extant networks. ireless networks such as WiMAX and Wi-Fi focus on providing perfection of service for VoIP. However, there are numerous aspects that affect quality of voice connections over wireless networks [13]. The adoption of Voice over Wireless Local Area
Network is on tremendous increase due its relief, non-invasive, economicexpansion, low maintenance cost, universal coverage and basic roaming capabilities. However, expansion Voice over Internet Protocol (VoIP) over Wireless Local Area Network (WLAN) is a challenging task for many network specialist and engineers. Voice codec is one of the most critical components of a VoIP system. In this project, we evaluate
the performance analysis of various codecs such as G.711, G.723 and G.729 over Wi-Fi networks. NS2 WiFi
simulation models are designed. Performance metrics such as Mean Opinion Score (MOS), average
end-to-end latency, and disconcert are evaluated and discussed [13]. 1. In this paper, our area of interest is to compare and study the performance analysis of VoIP codecs in Non-mobility scenarios by changing some parameters and plotting the graphs throughput, End to end Delay, MOS, Packet delivery Ratio, and Jitter by using Network Simulator version.
2. In this paper we analyze the different performance parameters, Recent research has focused on simulation studies with non- mobility scenarios to analyze different VoIP codecs with nodes up to 5. We have simulated the different VoIP codecs in non-mobility scenario with nodes up to 300.
CASE STUDY FOR PERFORMANCE ANALYSIS OF VOIP CODECS IN NON-MOBILITY SCENARIOSijcsity
IEEE 802.11 is the most popular standard for WLAN networks. It offers different physical transmission
rates. This paper focuses on this multi transmission rate of 802.11 WLANs and its effect on speech quality.
In non-adaptive systems, when the physical layer switches from a higher transmission rate to a lower one,
different than the one that the VoIP flow needs, the switching may result in congestion, high delay and
packet loss, and consequently speech quality degradation. However, there are some algorithms that adapt
the transmission parameters according to the channel conditions. In this study we demonstrate how
choosing parameter (different codec and packet size) can affect the voice quality, network delay and packet
loss. Further, this study presents a comparison between adaptive and non-adaptive methods. The adaptive method has also been evaluated for different congestion level from perceived speech quality point of view.
Improved voice quality with the combination of transport layer & audio codec ...journalBEEI
Improving voice quality over wireless communication becomes a demanding feature for social media apps like facebook, whatsapp and other communication channels. Voice-over-internet protocol (VoIP) helps us to make quick telephone calls over the internet. It includes various mechanism which are signaling, controlling and transport layer. Over wireless links, packet loss and high transmission delay damage voice quality. Here VoIP quality will be measured by three main elements which are signaling protocol, audio codec and transport layer. To improve the overall voice quality, we need to combine these three elements properly to get the best score. Otherwise perceptual speech quality will not be the right tool to measure the voice quality. Here we will use Mean Opinion Score (MOS) for calculated jitter values and end to end delay. At the end, best combination of audio codec & signaling protocol produced the quality speech.
Performance analysis of voip traffic over integrating wireless lan and wan us...ijwmn
A simulation model is presented to analyze and evaluate the performance of VoIP based integrated
wireless LAN/WAN with taking into account various voice encoding schemes. The network model was
simulated using OPNET Modeler software. Different parameters that indicate the QoS like MOS, jitter,
end to end delay, traffic send and traffic received are calculated and analyzed in Wireless LAN/WAN
scenarios. Depending on this evaluation, Selection codecs G.729A consider the best choice for VoIP.
A Comparative Analysis of the Performance of VoIP Traffic with Different Type...ijcnac
The key QoS parameters for VoIP are delay, jitter and loss. In the Internet, VoIP requires
the underlying packet switched network to minimize the impact of these parameters. A
major contributing factor in this regard is traffic engineering carried out by scheduling
algorithms. This paper studies the behavior of different types of scheduling algorithms on
the delay, jitter and loss QoS parameters. The performance evaluation involves
identifying the scheduling algorithms which are most suitable for VoIP communications.
The result from the analysis also shows the impact of the QoS parameters on VoIP over
the Internet.
Analyzing Video Streaming Quality by Using Various Error Correction Methods o...IJERA Editor
Transmission video over ad hoc networks has become one of the most important and interesting subjects of study for researchers and programmers because of the strong relationship between video applications and frequent users of various mobile devices, such as laptops, PDAs, and mobile phones in all aspects of life. However, many challenges, such as packet loss, congestion (i.e., impairments at the network layer), multipath fading (i.e., impairments at the physical layer) [1], and link failure, exist in transferring video over ad hoc networks; these challenges negatively affect the quality of the perceived video [2].This study has investigated video transfer over ad hoc networks. The main challenges of transferring video over ad hoc networks as well as types of errors that may occur during video transmission, various types of video mechanisms, error correction methods, and different Quality of Service (QoS) parameters that affect the quality of the received video are also investigated.
SERVICES AS PARAMETER TO PROVIDE BEST QOS : AN ANALYSIS OVER WIMAXijngnjournal
In this paper it is proposed to provide the QoS to the user by using the degradation of service under hostile environment being itself be a parameter to improve the QoS. Here the relation between the service and environment of its best performance drawn on the basis of simulation and analysis .The service then taken as a parameter to decide present environment of the user and to take measurable steps to improve the QoS either doing handover to nearby station or increasing power or to provide some marginal bandwidth etc.All analysis done over a WiMax network i.e. being designed and simulated using the Qualnet wireless simulator.
COMPARATIVE STUDY FOR PERFORMANCE ANALYSIS OF VOIP CODECS OVER WLAN IN NONMOB...Zac Darcy
Voice over IP (VoIP) applications such as Skype, Google Talk, and FaceTime are promising technologies for providing cheaper voice calls to end users over extant networks. ireless networks such as WiMAX and Wi-Fi focus on providing perfection of service for VoIP. However, there are numerous aspects that affect quality of voice connections over wireless networks [13]. The adoption of Voice over Wireless Local Area
Network is on tremendous increase due its relief, non-invasive, economicexpansion, low maintenance cost, universal coverage and basic roaming capabilities. However, expansion Voice over Internet Protocol (VoIP) over Wireless Local Area Network (WLAN) is a challenging task for many network specialist and engineers. Voice codec is one of the most critical components of a VoIP system. In this project, we evaluate
the performance analysis of various codecs such as G.711, G.723 and G.729 over Wi-Fi networks. NS2 WiFi
simulation models are designed. Performance metrics such as Mean Opinion Score (MOS), average
end-to-end latency, and disconcert are evaluated and discussed [13]. 1. In this paper, our area of interest is to compare and study the performance analysis of VoIP codecs in Non-mobility scenarios by changing some parameters and plotting the graphs throughput, End to end Delay, MOS, Packet delivery Ratio, and Jitter by using Network Simulator version.
2. In this paper we analyze the different performance parameters, Recent research has focused on simulation studies with non- mobility scenarios to analyze different VoIP codecs with nodes up to 5. We have simulated the different VoIP codecs in non-mobility scenario with nodes up to 300.
CASE STUDY FOR PERFORMANCE ANALYSIS OF VOIP CODECS IN NON-MOBILITY SCENARIOSijcsity
IEEE 802.11 is the most popular standard for WLAN networks. It offers different physical transmission
rates. This paper focuses on this multi transmission rate of 802.11 WLANs and its effect on speech quality.
In non-adaptive systems, when the physical layer switches from a higher transmission rate to a lower one,
different than the one that the VoIP flow needs, the switching may result in congestion, high delay and
packet loss, and consequently speech quality degradation. However, there are some algorithms that adapt
the transmission parameters according to the channel conditions. In this study we demonstrate how
choosing parameter (different codec and packet size) can affect the voice quality, network delay and packet
loss. Further, this study presents a comparison between adaptive and non-adaptive methods. The adaptive method has also been evaluated for different congestion level from perceived speech quality point of view.
Improved voice quality with the combination of transport layer & audio codec ...journalBEEI
Improving voice quality over wireless communication becomes a demanding feature for social media apps like facebook, whatsapp and other communication channels. Voice-over-internet protocol (VoIP) helps us to make quick telephone calls over the internet. It includes various mechanism which are signaling, controlling and transport layer. Over wireless links, packet loss and high transmission delay damage voice quality. Here VoIP quality will be measured by three main elements which are signaling protocol, audio codec and transport layer. To improve the overall voice quality, we need to combine these three elements properly to get the best score. Otherwise perceptual speech quality will not be the right tool to measure the voice quality. Here we will use Mean Opinion Score (MOS) for calculated jitter values and end to end delay. At the end, best combination of audio codec & signaling protocol produced the quality speech.
Performance analysis of voip traffic over integrating wireless lan and wan us...ijwmn
A simulation model is presented to analyze and evaluate the performance of VoIP based integrated
wireless LAN/WAN with taking into account various voice encoding schemes. The network model was
simulated using OPNET Modeler software. Different parameters that indicate the QoS like MOS, jitter,
end to end delay, traffic send and traffic received are calculated and analyzed in Wireless LAN/WAN
scenarios. Depending on this evaluation, Selection codecs G.729A consider the best choice for VoIP.
A Comparative Analysis of the Performance of VoIP Traffic with Different Type...ijcnac
The key QoS parameters for VoIP are delay, jitter and loss. In the Internet, VoIP requires
the underlying packet switched network to minimize the impact of these parameters. A
major contributing factor in this regard is traffic engineering carried out by scheduling
algorithms. This paper studies the behavior of different types of scheduling algorithms on
the delay, jitter and loss QoS parameters. The performance evaluation involves
identifying the scheduling algorithms which are most suitable for VoIP communications.
The result from the analysis also shows the impact of the QoS parameters on VoIP over
the Internet.
Analyzing Video Streaming Quality by Using Various Error Correction Methods o...IJERA Editor
Transmission video over ad hoc networks has become one of the most important and interesting subjects of study for researchers and programmers because of the strong relationship between video applications and frequent users of various mobile devices, such as laptops, PDAs, and mobile phones in all aspects of life. However, many challenges, such as packet loss, congestion (i.e., impairments at the network layer), multipath fading (i.e., impairments at the physical layer) [1], and link failure, exist in transferring video over ad hoc networks; these challenges negatively affect the quality of the perceived video [2].This study has investigated video transfer over ad hoc networks. The main challenges of transferring video over ad hoc networks as well as types of errors that may occur during video transmission, various types of video mechanisms, error correction methods, and different Quality of Service (QoS) parameters that affect the quality of the received video are also investigated.
SERVICES AS PARAMETER TO PROVIDE BEST QOS : AN ANALYSIS OVER WIMAXijngnjournal
In this paper it is proposed to provide the QoS to the user by using the degradation of service under hostile environment being itself be a parameter to improve the QoS. Here the relation between the service and environment of its best performance drawn on the basis of simulation and analysis .The service then taken as a parameter to decide present environment of the user and to take measurable steps to improve the QoS either doing handover to nearby station or increasing power or to provide some marginal bandwidth etc.All analysis done over a WiMax network i.e. being designed and simulated using the Qualnet wireless simulator.
Revamping quality of service of video streaming over wireless laneSAT Publishing House
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology
Performance analysis of voip over wired and wireless networks network impleme...eSAT Journals
Abstract In this Paper, the objective of simulation models is presented to investigate the performance of VoIP codecs over WiMAX and
FDDI networks that specially design for Aden University. To assure if the University IP network is prepared and adequate for this
new type of traffic before adding any new components, Aden University IP network will be simulated by using OPNET simulation
software then the new VoIP service will be added to the University networks. Different parameters that represent the QoS like end
to end delay, jitter, traffic sends and traffic received, MOS are calculated and analyzed in both network scenarios.
Keywords: VoIP, Codecs, QoS, WiMAX, FDDI
The Optimization of IPTV Service Through SDN In A MEC Architecture, Respectiv...CSCJournals
The aim of this paper is to present the ‘Power’ of SDN Technology and MEC Technic in improving the delivering of IPTV Service. Those days, the IPTV end –users are tremendous increased all over the world , but in the same time also the complains for receiving these prepaid real time multimedial services like; high latency, high bandwidth, low performance and low QoE/QoS. On the other end, IPTV Distributors need a new system, technics, network solutions to distribute content continuesly and simultaneously to all active end-users with high-quality, lowlatency and high Performance, thus monitoring and re-configuring this ‘Big Data’ require high Bandwidth by causing difficult problems by offering it affecting in the same time the price and QoE/QoSperformance of delivered service.
For this reason, we have achieved to optimize the IPTV service by applying SDN solution in a MEC Architecture (Multiple-Access Edge Computing). In this way , through MEC Technology and SDN, it is possible to receive an IPTV service with Low Latency, High Performance and Low Bandwidth by solving successfully all the problems faced by the actual IPTV Operators. These improvements of delivering IPTV service through MEC will be demonstrated by using the OMNet +++ simulator in an LTE-A mobile network. The results show clearly that by applying the MEC technique in the LTE-A network for receiving IPTV Service through SDN Network, the service was delivered with latency decreased by >90% (compared to the cases when the MEC technique is not applied), with PacketLoss of almost 0 and with high performance QoE. In addition these strong Contributions, the ‘Big’ innovation achieved in this work through simulations is that the quality of delivered IPTV Service did not change according to the increasing of the end-users.This latency of delivering the video streaming services did not change. This means that the IPTV Service providers will increase their benefits by ensuring in the same time also the delivering of service with high quality and performance toward innumerous end users. Consequently, MEC Technology and SDN solution will be the two right and "smart" network choices that will boost the development of the 5th Mobile generation and will significantly improve the benefit of Video Streaming services offered by current providers worldwide (Netflix, HULU, Amazon Prime, YouTube, etc).
Comparative Analysis of Quality of Service for Various Service Classes in WiM...Editor IJCATR
Broadband access is an important requirement to satisfy user demands and support a new set of real time services and
applications. WiMAX, as a Broadband Wireless Access solution for Wireless Metropolitan Area Networks, covering large distances
with high throughput and is a promising technology for Next Generation Networks. Nevertheless, for the successful deployment of
WiMAX based solutions, Quality of Service (QoS) is a mandatory feature that must be supported. Quality of Service (QoS) is an
important consideration for supporting variety of applications that utilize the network resources. These applications include voice over
IP, multimedia services, like, video streaming, video conferencing etc. In this paper the performances of the MPEG-4 High quality
video traffic over a WiMAX network using various service classes has been investigated. To analyze the QoS parameters, the WiMAX
module developed based on popular network simulator NS-3 is used. Various parameters that determine QoS of real life usage
scenarios and traffic flows of applications is analyzed. The objective is to compare different types of service classes with respect to the
QoS parameters, such as, throughput, packet loss, average delay and average jitter.
Over recent years there has been a considerable shift, from quality of service (QoS) to quality of experience (QoE), when evaluating video delivery across networks. Hence, we first explore the need for this shift towards user-QoE in the video delivery ecosystem. Further, we investigate major QoE metrics researchers use in the evaluation of DASH users. We point out a huge problem with DASH beginning with its transport layer protocol. DASH utilizes Transmission control protocol (TCP) as the transport layer protocol. Thus, we give an overview of the mechanism of Transmission Control Protocol (TCP) and two mechanisms greatly impacting the streaming process: (1) TCP
congestion mechanism and (2) TCP Fast Start. This leads us to investigate the impact of these TCP mechanisms on DASH players and consequently user-QoE.
QoS Based Capacity Enhancement for WCDMA Network with Coding SchemeVLSICS Design
The wide-band code division multiple access (WCDMA) based 3G and beyond cellular mobile wireless networks are expected to provide a diverse range of multimedia services to mobile users with guaranteed quality of service (QoS). To serve diverse quality of service requirements of these networks it necessitates new radio resource management strategies for effective utilization of network resources with coding schemes. Call admission control CAC) is a significant component in wireless networks to guarantee quality of service requirements and also to enhance the network resilience. In this paper capacity enhancement for WCDMA network with convolutional coding scheme is discussed and compared with block code and without coding scheme to achieve a better balance between resource utilization and quality of service provisioning. The model of this network is valid for the real-time (RT) and non-real-time (NRT) services having different data rate. Simulation results demonstrate the effectiveness of the network using convolutional code in terms of capacity enhancement and QoS of the voice and video services.
NETWORK PERFORMANCE EVALUATION WITH REAL TIME APPLICATION ENSURING QUALITY OF...ijngnjournal
The quality of service is a need in recent computer network developments. The present paper evaluates some characteristics in a proposed network topology such as dropped packets and bandwidth use, using two traffic sources, firstly a VoIP source over an UDP agent, then a CBR traffic source over an UDP agent as well as the previous one. Two possible configurations are proposed, implementing both of them in the Network Simulator, and implementing in one of them differentiated services to compare the results. Statistics results are shown, in both cases showing the accumulative dropped packet number and the throughput in the link, obtaining a reducer number of dropped packets in the stage with differentiated services, and an improvement in the bandwidth use.
Key management in information centric networkingIJCNCJournal
Information centric networking (ICN) has been in the spotlight of recent research. It is an emerging
communication paradigm that relays on the concept of publish and subscribe. It aims to revise the current
Internet with a new clean slate architecture where the design is completely different from today’s location
based model. To secure the forwarding plan in this network, it is vital to have a time based transient
forwarding identifiers by periodically changing the network link identifiers. This assumes shared keys to be
distributed prior the communications between an entity termed topology manager (TM) and each forwarder
in the network. Exchanging and sharing a secret key between two parties is one of most critical functions in
cryptography that needs to be more concerned when integrating cryptographic functions into the system. As
ICN is brand new Internet architecture, many existing cryptography protocols may need to be redesigned
to fit this new architecture. Therefore, this paper focuses on the security aspect of ICN and proposes an
initial design to deploy the integrated Diffie-Hellman-DSA key exchange protocol as a key distributions
mechanism.
Revamping quality of service of video streaming over wireless laneSAT Publishing House
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology
Performance analysis of voip over wired and wireless networks network impleme...eSAT Journals
Abstract In this Paper, the objective of simulation models is presented to investigate the performance of VoIP codecs over WiMAX and
FDDI networks that specially design for Aden University. To assure if the University IP network is prepared and adequate for this
new type of traffic before adding any new components, Aden University IP network will be simulated by using OPNET simulation
software then the new VoIP service will be added to the University networks. Different parameters that represent the QoS like end
to end delay, jitter, traffic sends and traffic received, MOS are calculated and analyzed in both network scenarios.
Keywords: VoIP, Codecs, QoS, WiMAX, FDDI
The Optimization of IPTV Service Through SDN In A MEC Architecture, Respectiv...CSCJournals
The aim of this paper is to present the ‘Power’ of SDN Technology and MEC Technic in improving the delivering of IPTV Service. Those days, the IPTV end –users are tremendous increased all over the world , but in the same time also the complains for receiving these prepaid real time multimedial services like; high latency, high bandwidth, low performance and low QoE/QoS. On the other end, IPTV Distributors need a new system, technics, network solutions to distribute content continuesly and simultaneously to all active end-users with high-quality, lowlatency and high Performance, thus monitoring and re-configuring this ‘Big Data’ require high Bandwidth by causing difficult problems by offering it affecting in the same time the price and QoE/QoSperformance of delivered service.
For this reason, we have achieved to optimize the IPTV service by applying SDN solution in a MEC Architecture (Multiple-Access Edge Computing). In this way , through MEC Technology and SDN, it is possible to receive an IPTV service with Low Latency, High Performance and Low Bandwidth by solving successfully all the problems faced by the actual IPTV Operators. These improvements of delivering IPTV service through MEC will be demonstrated by using the OMNet +++ simulator in an LTE-A mobile network. The results show clearly that by applying the MEC technique in the LTE-A network for receiving IPTV Service through SDN Network, the service was delivered with latency decreased by >90% (compared to the cases when the MEC technique is not applied), with PacketLoss of almost 0 and with high performance QoE. In addition these strong Contributions, the ‘Big’ innovation achieved in this work through simulations is that the quality of delivered IPTV Service did not change according to the increasing of the end-users.This latency of delivering the video streaming services did not change. This means that the IPTV Service providers will increase their benefits by ensuring in the same time also the delivering of service with high quality and performance toward innumerous end users. Consequently, MEC Technology and SDN solution will be the two right and "smart" network choices that will boost the development of the 5th Mobile generation and will significantly improve the benefit of Video Streaming services offered by current providers worldwide (Netflix, HULU, Amazon Prime, YouTube, etc).
Comparative Analysis of Quality of Service for Various Service Classes in WiM...Editor IJCATR
Broadband access is an important requirement to satisfy user demands and support a new set of real time services and
applications. WiMAX, as a Broadband Wireless Access solution for Wireless Metropolitan Area Networks, covering large distances
with high throughput and is a promising technology for Next Generation Networks. Nevertheless, for the successful deployment of
WiMAX based solutions, Quality of Service (QoS) is a mandatory feature that must be supported. Quality of Service (QoS) is an
important consideration for supporting variety of applications that utilize the network resources. These applications include voice over
IP, multimedia services, like, video streaming, video conferencing etc. In this paper the performances of the MPEG-4 High quality
video traffic over a WiMAX network using various service classes has been investigated. To analyze the QoS parameters, the WiMAX
module developed based on popular network simulator NS-3 is used. Various parameters that determine QoS of real life usage
scenarios and traffic flows of applications is analyzed. The objective is to compare different types of service classes with respect to the
QoS parameters, such as, throughput, packet loss, average delay and average jitter.
Over recent years there has been a considerable shift, from quality of service (QoS) to quality of experience (QoE), when evaluating video delivery across networks. Hence, we first explore the need for this shift towards user-QoE in the video delivery ecosystem. Further, we investigate major QoE metrics researchers use in the evaluation of DASH users. We point out a huge problem with DASH beginning with its transport layer protocol. DASH utilizes Transmission control protocol (TCP) as the transport layer protocol. Thus, we give an overview of the mechanism of Transmission Control Protocol (TCP) and two mechanisms greatly impacting the streaming process: (1) TCP
congestion mechanism and (2) TCP Fast Start. This leads us to investigate the impact of these TCP mechanisms on DASH players and consequently user-QoE.
QoS Based Capacity Enhancement for WCDMA Network with Coding SchemeVLSICS Design
The wide-band code division multiple access (WCDMA) based 3G and beyond cellular mobile wireless networks are expected to provide a diverse range of multimedia services to mobile users with guaranteed quality of service (QoS). To serve diverse quality of service requirements of these networks it necessitates new radio resource management strategies for effective utilization of network resources with coding schemes. Call admission control CAC) is a significant component in wireless networks to guarantee quality of service requirements and also to enhance the network resilience. In this paper capacity enhancement for WCDMA network with convolutional coding scheme is discussed and compared with block code and without coding scheme to achieve a better balance between resource utilization and quality of service provisioning. The model of this network is valid for the real-time (RT) and non-real-time (NRT) services having different data rate. Simulation results demonstrate the effectiveness of the network using convolutional code in terms of capacity enhancement and QoS of the voice and video services.
NETWORK PERFORMANCE EVALUATION WITH REAL TIME APPLICATION ENSURING QUALITY OF...ijngnjournal
The quality of service is a need in recent computer network developments. The present paper evaluates some characteristics in a proposed network topology such as dropped packets and bandwidth use, using two traffic sources, firstly a VoIP source over an UDP agent, then a CBR traffic source over an UDP agent as well as the previous one. Two possible configurations are proposed, implementing both of them in the Network Simulator, and implementing in one of them differentiated services to compare the results. Statistics results are shown, in both cases showing the accumulative dropped packet number and the throughput in the link, obtaining a reducer number of dropped packets in the stage with differentiated services, and an improvement in the bandwidth use.
Key management in information centric networkingIJCNCJournal
Information centric networking (ICN) has been in the spotlight of recent research. It is an emerging
communication paradigm that relays on the concept of publish and subscribe. It aims to revise the current
Internet with a new clean slate architecture where the design is completely different from today’s location
based model. To secure the forwarding plan in this network, it is vital to have a time based transient
forwarding identifiers by periodically changing the network link identifiers. This assumes shared keys to be
distributed prior the communications between an entity termed topology manager (TM) and each forwarder
in the network. Exchanging and sharing a secret key between two parties is one of most critical functions in
cryptography that needs to be more concerned when integrating cryptographic functions into the system. As
ICN is brand new Internet architecture, many existing cryptography protocols may need to be redesigned
to fit this new architecture. Therefore, this paper focuses on the security aspect of ICN and proposes an
initial design to deploy the integrated Diffie-Hellman-DSA key exchange protocol as a key distributions
mechanism.
A review study of handover performance in mobile ipIJCNCJournal
The Mobile Internet Protocol (Mobile IP) is an extension to the Internet Protocol proposed by the Internet
Engineering Task Force (IETF) that addresses the mobility issues. In order to support un-interrupted
services and seamless mobility of nodes across the networks (and/or sub-networks) with permanent IP
addresses, handover is performed in mobile IP enabled networks. Handover in mobile IP is source cause of
performance degradation as it results in increased latency and packet loss during handover. Other issues
like scalability issues, ordered packet delivery issues, control plane management issues etc are also
adversely affected by it. The paper provides a constructive survey by classifying, discussing and comparing
different handover techniques that have been proposed so far, for enhancing the performance during
handovers. Finally some general solutions that have been used to solve handover related problems are
briefly discussed.
Framework for wireless network security using quantum cryptographyIJCNCJournal
Data that is transient over an unsecured wireless network is always susceptible to being intercepted by
anyone within the range of the wireless signal. Hence providing secure communication to keep the user’s
information and devices safe when connected wirelessly has become one of the major concerns. Quantum cryptography provides a solution towards absolute communication security over the network by encoding
information as polarized photons, which can be sent through the air. This paper explores on the aspect of
application of quantum cryptography in wireless networks.
In this paper we present a methodology for integrating quantum cryptography and security of IEEE 802.11 wireless networks in terms of distribution of the encryption keys.
A novel scheme to improve the spectrum sensing performanceIJCNCJournal
Due to limited availability of spectrum for license
d users only, the need for secondary access by unli
censed
users is increasing. Cognitive radio turns out to b
e helping this situation because all that is needed
is a
technique that could efficiently detect the empty s
paces and provide them to the secondary devices wit
hout
causing any interference to the primary (licensed)
users. Spectrum sensing is the foremost function of
the
cognitive radio which senses the environment for wh
ite spaces. Energy detection is one of the various
spectrum sensing techniques that are under research
. Earlier it was shown that energy detection works
better under AWGN channel as compared to Rayleigh c
hannel, however the conventional spectrum sensing
techniques have a high probability of false alarm a
nd also show a better probability of detection for
higher
values of SNR. There is a need for a new technique
that shows a reduced probability of false alarm as
well
as an increase in the probability of detection for
lower values of SNR. In the present work the conven
tional
energy detection technique has been enhanced to get
better results.
Different date block size using to evaluate the performance between different...IJCNCJournal
The different computer networks whether wired or wireless are becoming more popular with its high
security aspect. Different security algorithms and technique are using to avoid any aforementioned attacks.
One of these technique is a cryptography technique that makes the data as unreadable during the transfer
hence; there is no chance to reclaim the information. Presently, most of the users are using various media
types and internet to transfer the data but, it has the chance to retrieve the data by using these media types.
The perfect solution for this problem is to provide security on time-to-time basis; this stage is always
significant to the security related community discussions. This paper explains the comparison between the
run time of three different encryption algorithms which are DES, AES and Blowfish The compression
includes using different modes, data block size and different operation modes. As a result, Blowfish
algorithm followed by AES take less time for running compared to DES.
On the development of methodology for planning and cost modeling of a wide ar...IJCNCJournal
The most important stages in designing a
computer
network
in a
wider geographical area include:
definition of requirements, topological description
,
identification and calculation of relevant parameters
(
i
.
e
.
traffic matrix
)
, determining the shortest path between nodes, quantification of the effect of various
levels
of technical and technological development of urban areas involved, the cost of technology
,
and the
cost of services. The
se
parameters differ for WAN networks in different regions
–
their calculation depends
directly
on
the data “
i
n the field
”
: number of inhabitants, distance between populated areas,
network
traffic
density
,
as well as
available
bandwidth
. The
main
reason for identification and evaluation of these
parameters
is
to develop a model that could
meet the
constraints
im
posed by poten
tial beneficiaries.
In this
paper
,
we develop a methodology for planning and cost
-
modeling of a wide area network
and
validate it
in
a case study,
under the
supposition
that
behavioral interactions of individuals and groups play a significant
role and have
to be taken into consideration
by employing either simple or composite indicators of
socioeconomic status
.
Dcf learn and performance analysis of 802.11 b wireless networkIJCNCJournal
Though WLAN wireless network has been widely deployed as the main split-flow deployment of the
communication network, little study emphasizes its performance as WLAN protocols were only designed for
the public communicating conveniently with each other. Actually that too much wireless access points
assembling together will cause self-interference to the whole WLAN network. This paper investigates the
distributed coordination function (DCF) learn and the performance study of 802.11b networks. Firstly, our
study illustrates the performance of its MAC layer and its fairness issues related to DCF. Next we propose
the details which should be paid attention to in deploying network services. Then, performance analyses
are evaluated by simulation and real test for a dense wireless network. Our main goal is to give proposals
to network operators how to design a WLAN network more standardized and orderly.
A fuzzy logic controllerfora two link functional manipulatorIJCNCJournal
This paper presents a new approach for designing a Fuzzy Logic Controller "FLC"for a dynamically multivariable nonlinear coupling system. The conventional controller with constant gains for different operating points may not be sufficient to guarantee satisfactory performance for Robot manipulator. The Fuzzy Logic Controller utilizes the error and the change of error as fuzzy linguistic inputs to regulate the system performance. The proposed controller have been developed to simulate the dynamic behavior of A
Two-Link Functional Manipulator. The new controller uses only the available information of the input-output for controlling the position and velocity of the robot axes of the motion of the end effectors
PAPR REDUCTION OF OFDM SIGNAL BY USING COMBINED HADAMARD AND MODIFIED MEU-LAW...IJCNCJournal
Orthogonal frequency division multiplexing (OFDM) is a technique which gives high quality of service (QOS) to the users by mitigating the fading signals as well as high data rates in multimedia services. However, the peak-to-average power ratio (PAPR) is a technical challenge that reduces the efficiency of RF power amplifiers. In this paper, we propose the combined Hadamard transform and modified meu-law companding transform method in order to lessen the effects of the peak-to-average power ratio of the
OFDM signal. Simulation results show that the proposed scheme reduces PAPR compared to other companding techniques as well as the Hadamard transform technique when used on its own.
LTE QOS DYNAMIC RESOURCE BLOCK ALLOCATION WITH POWER SOURCE LIMITATION AND QU...IJCNCJournal
3GPP has defined the long term evolution (LTE) for 3G radio access in order to maintain the future
competitiveness for 3G technology, the system provides the capability of supporting a mixture of services
with different quality of service (QoS) requirements. This paper proposes a new cross-layer scheduling
algorithm to satisfy better QoS parameters for real time applications. The proposed algorithm takes care of
allocating resource blocks (RBs) with different modulation and coding schemes (MCS) according to target
bit error rate (BER), user equipment supportable MCS, queue stability constraints and available transmit
power constraints. The proposed algorithm has been valued, compared with an earlier allocation algorithm
in terms of service rate and packet delay and showed better performance regards the real time
applications.
Further results on the joint time delay and frequency estimation without eige...IJCNCJournal
Joint Time Delay and Frequency Estimation (JTDFE) problem of complex sinusoidal signals received at
two separated sensors is an attractive problem that has been considered for several engineering
applications. In this paper, a high resolution null (noise) subspace method without eigenvalue
decomposition is proposed. The direct data Matrix is replaced by an upper triangular matrix obtained from
Rank-Revealing LU (RRLU) factorization. The RRLU provides accurate information about the rank and the
numerical null space which make it a valuable tool in numerical linear algebra.The proposed novel method
decreases the computational complexity of JTDFE approximately to the half compared with RRQR
methods. The proposed method generates estimates of the unknown parameters which are based on the
observation and/or covariance matrices. This leads to a significant improvement in the computational load.
Computer simulations are included in this paper to demonstrate the proposed method.
ETOR-Efficient Token based Opportunistic RoutingIJCNCJournal
This paper proposes an Efficient Token based Opportunistic Routing called ETOR, which is an
improvement to the token based coordination approach for opportunistic routing proposed by Economy[1].
In Economy, method used for finding the connected candidate order chooses neighbor as the next
candidate by considering ETX of that neighbor towards the source but it does not consider the link
probability between the relay candidate and neighbor to be selected. ETOR proposes variant methods for
finding the connected candidate order in token based opportunistic routing by considering both the ETX
of the neighbor towards source as well as ETX of the relay towards sending candidate which avoids weaker
links between its intermediate nodes thereby improving the throughput and reducing the AA Ratio. We also
propose a solution for reducing the number of hops traversed by the token, which in turn increases the
token generation speed. Simulation results show that the proposed ETOR approaches perform better than
Economy approach in terms of AA Ratio, number of hops traversed by the token and number of token
traversals.
Comparative Study for Performance Analysis of VOIP Codecs Over WLAN in Nonmob...Zac Darcy
Voice over IP (VoIP) applications such as Skype, Google Talk, and FaceTime are promising technologies for providing cheaper voice calls to end users over extant networks. Wireless networks such as WiMAX and Wi-Fi focus on providing perfection of service for VoIP. However, there are numerous aspects that affect quality of voice connections over wireless networks [13]. The adoption of Voice over Wireless Local Area Network is on tremendous increase due its relief, non-invasive, economicexpansion, low maintenance cost, universal coverage and basic roaming capabilities.
Comparative Study for Performance Analysis of VOIP Codecs Over WLAN in Nonmob...Zac Darcy
Voice over IP (VoIP) applications such as Skype, Google Talk, and FaceTime are promising technologies
for providing cheaper voice calls to end users over extant networks. Wireless networks such as WiMAX and
Wi-Fi focus on providing perfection of service for VoIP. However, there are numerous aspects that affect
quality of voice connections over wireless networks [13]. The adoption of Voice over Wireless Local Area
Network is on tremendous increase due its relief, non-invasive, economicexpansion, low maintenance cost,
universal coverage and basic roaming capabilities. However, expansion Voice over Internet Protocol
(VoIP) over Wireless Local Area Network (WLAN) is a challenging task for many network specialist and
engineers. Voice codec is one of the most critical components of a VoIP system. In this project, we evaluate
the performance analysis of various codecs such as G.711, G.723 and G.729 over Wi-Fi networks. NS2 WiFi simulation models are designed. Performance metrics such as Mean Opinion Score (MOS), average
end-to-end latency, and disconcert are evaluated and discussed [13].
1. In this paper, our area of interest is to compare and study the performance analysis of VoIP
codecs in Non-mobility scenarios by changing some parameters and plotting the graphs
throughput, End to end Delay, MOS, Packet delivery Ratio, and Jitter by using Network
Simulator version.
2. In this paper we analyze the different performance parameters, Recent research has focused on
simulation studies with non- mobility scenarios to analyze different VoIP codecs with nodes up to
5. We have simulated the different VoIP codecs in non-mobility scenario with nodes up to 300.
COMPARATIVE STUDY FOR PERFORMANCE ANALYSIS OF VOIP CODECS OVER WLAN IN NONMOB...Zac Darcy
The recent Voice over IP (VOIP) applications such as Skype, Google Talk, and Face Time have
changed the way people communicate to each other. Due to the low cost, people find VOIP as an
alternative to the expensive traditional Public Switched Telephone Network (PSTN). VOIP has
set of parameters that defined its Quality of Service (QoS) such as end to delay, jitter, packets
loss, Mean Opinion Score (MOS, and throughput[13]. The existing wireless networks such as WiFi offer flexibility to support such applications. At the time the IEEE 802.11 (Wi-Fi) technology
showed great success as cheap wireless internet access. The Motive of this survey paper is to
analyse of Qos in VOIP [13].
COMPARATIVE STUDY FOR PERFORMANCE ANALYSIS OF VOIP CODECS OVER WLAN IN NONMOB...Zac Darcy
Voice over IP (VoIP) applications such as Skype, Google Talk, and FaceTime are promising technologies
for providing cheaper voice calls to end users over extant networks. Wireless networks such as WiMAX and
Wi-Fi focus on providing perfection of service for VoIP. However, there are numerous aspects that affect
quality of voice connections over wireless networks [13]. The adoption of Voice over Wireless Local Area
Network is on tremendous increase due its relief, non-invasive, economicexpansion, low maintenance cost,
universal coverage and basic roaming capabilities. However, expansion Voice over Internet Protocol
(VoIP) over Wireless Local Area Network (WLAN) is a challenging task for many network specialist and
engineers. Voice codec is one of the most critical components of a VoIP system. In this project, we evaluate
the performance analysis of various codecs such as G.711, G.723 and G.729 over Wi-Fi networks. NS2 WiFi simulation models are designed. Performance metrics such as Mean Opinion Score (MOS), average
end-to-end latency, and disconcert are evaluated and discussed [13]
SPEECH QUALITY EVALUATION BASED CODEC FOR VOIP OVER 802.11P ijwmn
Voice over Internet Protocol (VoIP) may provide good services through Vehicular ad hoc networks
(VANETs) platform by providing services to many application scenarios range from safety to comfort.
However, VANETs networks introduce many challenges for supporting voice with QoS requirements. In
this paper, our study is based on Inter-Vehicle voice streaming rely on multi-hop fashion. For this task, a
performance evaluation of various audio CODECs will be analyzed by mean of simulations.
Furthermore, we test the impact of network environment on QoS metrics. To achieve good results,
CODECs behaviour is tested by using mobility information obtained from vehicular traffic generator. The
mobility model is based on the real road maps of an urban environment. Focusing on inter-vehicular
voice traffic quality, we provide simulations results in terms of both user level (MOS) metrics and
network level (such as Losses). According to this performance evaluation, we show that G.723.1 CODEC
worked well in the urban VANET environment.
Analysis of VoIP Traffic in WiMAX EnvironmentEditor IJMTER
Worldwide Interoperability for Microwave Access (WiMAX) is currently one of the
hottest technologies in wireless communication. It is a standard based on the IEEE 802.16 wireless
technology that provides a very high throughput broadband connections over long distances. In
parallel, Voice Over Internet Protocol (VoIP) is a new technology which provides access to voice
communication over internet protocol and hence it is becomes an alternative to public switched
telephone networks (PSTN) due to its capability of transmission of voice as packets over IP
networks. A lot of research has been done in analyzing the performances of VoIP traffic over
WiMAX network. In this paper we review the analysis carried out by several authors for the most
common VoIP codec’s which are G.711, G.723.1 and G.729 over a WiMAX network using various
service classes. The objective is to compare the results for different types of service classes with
respect to the QoS parameters such as throughput, average delay and average jitter.
VOIP PERFORMANCE OVER BROADBAND WIRELESS NETWORKS UNDER STATIC AND MOBILE ENV...ijwmn
Voice over IP is expected to be very promising application in the next generation communication networks. The objective of this paper is to analyse the VoIP performance among the most competing next generation wireless networks like WiMAX, WLAN and its integrated frameworks etc. WiMAX having higher bandwidth provides higher capacity but with degraded quality of service while WLAN provides low capacity and coverage. Hence, an integrated network using WiMAX backbone and WLAN hotspots has been developed and VoIP application has been setup. As OPNET 14.5.A provides a real life simulation environment, we have opted OPNET as the simulation platform for all performance studies in this work. Quality of the service is critically analysed with parameters like jitter, MOS and delay for various voice codecs in the aforesaid networks for both fixed and mobile scenario. Finally, it is observed and concluded that the WiMAX-WLAN integrated network provides improved and optimal performance over WLAN and WiMAX network with respect to network capacity and quality of service. Exhaustive simulation results have been provided.
Throughput Performance Analysis VOIP over LTEiosrjce
IOSR Journal of Electronics and Communication Engineering(IOSR-JECE) is a double blind peer reviewed International Journal that provides rapid publication (within a month) of articles in all areas of electronics and communication engineering and its applications. The journal welcomes publications of high quality papers on theoretical developments and practical applications in electronics and communication engineering. Original research papers, state-of-the-art reviews, and high quality technical notes are invited for publications.
Comparisons of QoS in VoIP over WIMAX by Varying the Voice codes and Buffer sizeEditor IJCATR
Voice over Internet Protocol (VoIP) is developed for voice communications system based on voice packets transmitted over
IP network with real-time communications of voice across networks using the Internet protocols. Quality of Service (QoS) mechanism
is applied to guarantee successful voice packets transmitted over IP network with reduced delay or drop according to assigned priority
of voice packets. In this paper, the goal of simulation models is present to investigate the performance of VoIP codecs and buffer size
for improving quality of service (QoS) with the simulation results by using OPNET modeler version 14.5. The performance of the
proposed algorithm is analyzed and compared the quality of service for VoIP. The final simulated result shows that the VoIP service
performance best under G.729 voice encoder scheme and buffer size 256 Kb over WiMAX network.
The paper discussed the implementation of the use of
Voice over Internet Protocol services, which is resulting in the
enormous growth of broadband network. The main objective
of this paper is to evaluate the QoS of VOIP for different
broad band networks. Wired, Wireless Local Area Network
(WLAN), Worldwide Interoperability for Microwave Access
(WiMAX) and Universal Mobile Telecommunication System
(UMTS) networks were implemented in OPNET Modeler The
quality is compared using different QoS parameters such as
end-to-end delay, Mean Opion Score (MOS), throughput and
jitter. The VoIP codes used in the measurements of QoS is
G.729A with sampling rate of 8 kbps .The results analyzed and
the performance evaluated will give network operators an
opportunity to select the codec for better services of VoIP for
customer satisfaction.
PERFORMANCE EVALUATION OF OSPF AND RIP ON IPV4 & IPV6 TECHNOLOGY USING G.711 ...IJCNCJournal
Migration from IPv4 to IPv6 is still visibly slow, mainly because of the inherent cost involved in the implementation, hardware and software acquisition. However, there are many values IPv6 can bring to the
IP enabled environment as compared to IPv4, particularly for Voice Over Internet Protocol (VoIP) solutions. Many companies are drifting away from circuit based switching such as PSTN to packet based switching (VoIP) for collaboration. There are several factors determining the effective utilization and
quality of VoIP solutions. These include the choice of codec, echo control, packet loss, delay, delay variation (jitter), and the network topology. The network is basically the environment in which VoIP is deployed. State of art network design for VoIP technologies requires impeccable Interior Gateway routing
protocols that will reduce the convergence time of the network, in the event of a link failure. Choice of CODEC is also a main factor. Since most research work in this area did not consider a particular CODEC as a factor in determining performance, this paper will compare the behaviour of RIP and OSPF in IPv4
and IPv6 using G.711 CODEC with riverbed modeller17.5.
PERFORMANCE EVALUATION OF OSPF AND RIP ON IPV4 & IPV6 TECHNOLOGY USING G.711 ...IJCNCJournal
Migration from IPv4 to IPv6 is still visibly slow, mainly because of the inherent cost involved in the implementation, hardware and software acquisition. However, there are many values IPv6 can bring to the
IP enabled environment as compared to IPv4, particularly for Voice Over Internet Protocol (VoIP) solutions. Many companies are drifting away from circuit based switching such as PSTN to packet based switching (VoIP) for collaboration. There are several factors determining the effective utilization and
quality of VoIP solutions. These include the choice of codec, echo control, packet loss, delay, delay variation (jitter), and the network topology. The network is basically the environment in which VoIP is deployed. State of art network design for VoIP technologies requires impeccable Interior Gateway routing
protocols that will reduce the convergence time of the network, in the event of a link failure. Choice of CODEC is also a main factor. Since most research work in this area did not consider a particular CODEC as a factor in determining performance, this paper will compare the behaviour of RIP and OSPF in IPv4
and IPv6 using G.711 CODEC with riverbed modeller17.5.
Video steaming Throughput Performance Analysis over LTEiosrjce
IOSR Journal of Electronics and Communication Engineering(IOSR-JECE) is a double blind peer reviewed International Journal that provides rapid publication (within a month) of articles in all areas of electronics and communication engineering and its applications. The journal welcomes publications of high quality papers on theoretical developments and practical applications in electronics and communication engineering. Original research papers, state-of-the-art reviews, and high quality technical notes are invited for publications.
Vehicle Ad Hoc Networks (VANETs) have become a viable technology to improve traffic flow and safety on the roads. Due to its effectiveness and scalability, the Wingsuit Search-based Optimised Link State Routing Protocol (WS-OLSR) is frequently used for data distribution in VANETs. However, the selection of MultiPoint Relays (MPRs) plays a pivotal role in WS-OLSR's performance. This paper presents an improved MPR selection algorithm tailored to WS-OLSR, designed to enhance the overall routing efficiency and reduce overhead. The analysis found that the current OLSR protocol has problems such as redundancy of HELLO and TC message packets or failure to update routing information in time, so a WS-OLSR routing protocol based on improved-MPR selection algorithm was proposed. Firstly, factors such as node mobility and link changes are comprehensively considered to reflect network topology changes, and the broadcast cycle of node HELLO messages is controlled through topology changes. Secondly, a new MPR selection algorithm is proposed, considering link stability issues and nodes. Finally, evaluate its effectiveness in terms of packet delivery ratio, end-to-end delay, and control message overhead. Simulation results demonstrate the superior performance of our improved MR selection algorithm when compared to traditional approaches.
A Novel Medium Access Control Strategy for Heterogeneous Traffic in Wireless ...IJCNCJournal
So far, Wireless Body Area Networks (WBANs) have played a pivotal role in driving the development of intelligent healthcare systems with broad applicability across various domains. Each WBAN consists of one or more types of sensors that can be embedded in clothing, attached directly to the body, or even implanted beneath an individual's skin. These sensors typically serve asingle application. However, the traffic generated by each sensor may have distinct requirements. This diversity necessitates a dual approach: tailored treatment based on the specific needs of each traffic typeand the fulfillment of application requirements, such asreliability and timeliness. Never the less, the presence of energy constraints and the unreliable nature of wireless communications make QoS provisioning under such networks a non-trivial task. In this context, the current paper introduces a novel Medium AccessControl (MAC) strategy for the regular traffic applications of WBANs, designed to significantly enhance efficiency when compared to the established MAC protocols IEEE 802.15.4 and IEEE 802.15.6, with a particular focus on improving reliability, timeliness, and energy efficiency.
May_2024 Top 10 Read Articles in Computer Networks & Communications.pdfIJCNCJournal
The International Journal of Computer Networks & Communications (IJCNC) is a bi monthly open access peer-reviewed journal that publishes articles which contribute new results in all areas of Computer Networks & Communications. The journal focuses on all technical and practical aspects of Computer Networks & data Communications. The goal of this journal is to bring together researchers and practitioners from academia and industry to focus on advanced networking concepts and establishing new collaborations in these areas.
A Topology Control Algorithm Taking into Account Energy and Quality of Transm...IJCNCJournal
The efficient use of energy in wireless sensor networks is critical for extending node lifetime. The network topology is one of the factors that have a significant impact on the energy usage at the nodes and the quality of transmission (QoT) in the network. We propose a topology control algorithm for software-defined wireless sensor networks (SDWSNs) in this paper. Our method is to formulate topology control algorithm as a nonlinear programming (NP) problem with the objective to optimizing two metrics, maximum communication range, and desired degree. This NP problem is solved at the SDWSN controller by employing the genetic algorithm (GA) to determine the best topology. The simulation results show that the proposed algorithm outperforms the MaxPower algorithm in terms of average node degree and energy expansion ratio.
Multi-Server user Authentication Scheme for Privacy Preservation with Fuzzy C...IJCNCJournal
The integration of artificial intelligence technology with a scalable Internet of Things (IoT) platform facilitates diverse smart communication services, allowing remote users to access services from anywhere at any time. The multi-server environment within IoT introduces a flexible security service model, enabling users to interact with any server through a single registration. To ensure secure and privacy preservation services for resources, an authentication scheme is essential. Zhao et al. recently introduced a user authentication scheme for the multi-server environment, utilizing passwords and smart cards, claiming resilience against well-known attacks. This paper conducts cryptanalysis on Zhao et al.'s scheme, focusing on denial of service and privacy attacks, revealing a lack of user-friendliness. Subsequently, we propose a new multi-server user authentication scheme for privacy preservation with fuzzy commitment over the IoT environment, addressing the shortcomings of Zhao et al.'s scheme. Formal security verification of the proposed scheme is conducted using the ProVerif simulation tool. Through both formal and informal security analyses, we demonstrate that the proposed scheme is resilient against various known attacks and those identified in Zhao et al.'s scheme.
Advanced Privacy Scheme to Improve Road Safety in Smart Transportation SystemsIJCNCJournal
In -Vehicle Ad-Hoc Network (VANET), vehicles continuously transmit and receive spatiotemporal data with neighboring vehicles, thereby establishing a comprehensive 360-degree traffic awareness system. Vehicular Network safety applications facilitate the transmission of messages between vehicles that are near each other, at regular intervals, enhancing drivers' contextual understanding of the driving environment and significantly improving traffic safety. Privacy schemes in VANETs are vital to safeguard vehicles’ identities and their associated owners or drivers. Privacy schemes prevent unauthorized parties from linking the vehicle's communications to a specific real-world identity by employing techniques such as pseudonyms, randomization, or cryptographic protocols. Nevertheless, these communications frequently contain important vehicle information that malevolent groups could use to Monitor the vehicle over a long period. The acquisition of this shared data has the potential to facilitate the reconstruction of vehicle trajectories, thereby posing a potential risk to the privacy of the driver. Addressing the critical challenge of developing effective and scalable privacy-preserving protocols for communication in vehicle networks is of the highest priority. These protocols aim to reduce the transmission of confidential data while ensuring the required level of communication. This paper aims to propose an Advanced Privacy Vehicle Scheme (APV) that periodically changes pseudonyms to protect vehicle identities and improve privacy. The APV scheme utilizes a concept called the silent period, which involves changing the pseudonym of a vehicle periodically based on the tracking of neighboring vehicles. The pseudonym is a temporary identifier that vehicles use to communicate with each other in a VANET. By changing the pseudonym regularly, the APV scheme makes it difficult for unauthorized entities to link a vehicle's communications to its real-world identity. The proposed APV is compared to the SLOW, RSP, CAPS, and CPN techniques. The data indicates that the efficiency of APV is a better improvement in privacy metrics. It is evident that the AVP offers enhanced safety for vehicles during transportation in the smart city.
April 2024 - Top 10 Read Articles in Computer Networks & CommunicationsIJCNCJournal
The International Journal of Computer Networks & Communications (IJCNC) is a bi monthly open access peer-reviewed journal that publishes articles which contribute new results in all areas of Computer Networks & Communications. The journal focuses on all technical and practical aspects of Computer Networks & data Communications. The goal of this journal is to bring together researchers and practitioners from academia and industry to focus on advanced networking concepts and establishing new collaborations in these areas.
DEF: Deep Ensemble Neural Network Classifier for Android Malware DetectionIJCNCJournal
Malware is one of the threats to security of computer networks and information systems. Since malware instances are available sufficiently, there is increased interest among researchers on usage of Artificial Intelligence (AI). Of late AI-enabled methods such as machine learning (ML) and deep learning paved way for solving many real-world problems. As it is a learning-based approach, accumulated training samples help in improving thequality of training and thus leveraging malware detection accuracy. Existing deep learning methods are focusing on learning-based malware detection systems. However, there is need for improving the state of the art through ensemble approach. Towards this end, in this paper we proposed a framework known as Deep Ensemble Framework (DEF) for automatic malware detection. The framework obtains features from training samples. From given malware instance a grayscale image is generated. There is another process to extract the opcode sequences. Convolutional Neural Network (CNN) and Long Short Term Memory (LSTM) techniques are used to obtain grayscale image and opcode sequence respectively. Afterwards, a stacking ensemble is employed in order to achieve efficient malware detection and classification. Malware samples collected fromthe Internet sources and Microsoft are used for theempirical study. An algorithm known as Ensemble Learning for Automatic Malware Detection (EL-AML) is proposed to realize our framework. Another algorithm named Pre-Process is proposed to assist the EL-AML algorithm for obtaining intermediate features required by CNN and LSTM.Empirical study reveals that our framework outperforms many existing methods in terms of speed-up and accuracy.
High Performance NMF Based Intrusion Detection System for Big Data IOT TrafficIJCNCJournal
With the emergence of smart devices and the Internet of Things (IoT), millions of users connected to the network produce massive network traffic datasets. These vast datasets of network traffic, Big Data are challenging to store, deal with and analyse using a single computer. In this paper we developed parallel implementation using a High Performance Computer (HPC) for the Non-Negative Matrix Factorization technique as an engine for an Intrusion Detection System (HPC-NMF-IDS). The large IoT traffic datasets of order of millions samples are distributed evenly on all the computing cores for both storage and speedup purpose. The distribution of computing tasks involved in the Matrix Factorization takes into account the reduction of the communication cost between the computing cores. The experiments we conducted on the proposed HPC-IDS-NMF give better results than the traditional ML-based intrusion detection systems. We could train the HPC model with datasets of one million samples in only 31 seconds instead of the 40 minutes using one processor), that is a speed up of 87 times. Moreover, we have got an excellent detection accuracy rate of 98% for KDD dataset.
A Novel Medium Access Control Strategy for Heterogeneous Traffic in Wireless ...IJCNCJournal
So far, Wireless Body Area Networks (WBANs) have played a pivotal role in driving the development of intelligent healthcare systems with broad applicability across various domains. Each WBAN consists of one or more types of sensors that can be embedded in clothing, attached directly to the body, or even implanted beneath an individual's skin. These sensors typically serve asingle application. However, the traffic generated by each sensor may have distinct requirements. This diversity necessitates a dual approach: tailored treatment based on the specific needs of each traffic typeand the fulfillment of application requirements, such asreliability and timeliness. Never the less, the presence of energy constraints and the unreliable nature of wireless communications make QoS provisioning under such networks a non-trivial task. In this context, the current paper introduces a novel Medium AccessControl (MAC) strategy for the regular traffic applications of WBANs, designed to significantly enhance efficiency when compared to the established MAC protocols IEEE 802.15.4 and IEEE 802.15.6, with a particular focus on improving reliability, timeliness, and energy efficiency.
A Topology Control Algorithm Taking into Account Energy and Quality of Transm...IJCNCJournal
The efficient use of energy in wireless sensor networks is critical for extending node lifetime. The network topology is one of the factors that have a significant impact on the energy usage at the nodes and the quality of transmission (QoT) in the network. We propose a topology control algorithm for software-defined wireless sensor networks (SDWSNs) in this paper. Our method is to formulate topology control algorithm as a nonlinear programming (NP) problem with the objective to optimizing two metrics, maximum communication range, and desired degree. This NP problem is solved at the SDWSN controller by employing the genetic algorithm (GA) to determine the best topology. The simulation results show that the proposed algorithm outperforms the MaxPower algorithm in terms of average node degree and energy expansion ratio.
Multi-Server user Authentication Scheme for Privacy Preservation with Fuzzy C...IJCNCJournal
The integration of artificial intelligence technology with a scalable Internet of Things (IoT) platform facilitates diverse smart communication services, allowing remote users to access services from anywhere at any time. The multi-server environment within IoT introduces a flexible security service model, enabling users to interact with any server through a single registration. To ensure secure and privacy preservation services for resources, an authentication scheme is essential. Zhao et al. recently introduced a user authentication scheme for the multi-server environment, utilizing passwords and smart cards, claiming resilience against well-known attacks. This paper conducts cryptanalysis on Zhao et al.'s scheme, focusing on denial of service and privacy attacks, revealing a lack of user-friendliness. Subsequently, we propose a new multi-server user authentication scheme for privacy preservation with fuzzy commitment over the IoT environment, addressing the shortcomings of Zhao et al.'s scheme. Formal security verification of the proposed scheme is conducted using the ProVerif simulation tool. Through both formal and informal security analyses, we demonstrate that the proposed scheme is resilient against various known attacks and those identified in Zhao et al.'s scheme.
Advanced Privacy Scheme to Improve Road Safety in Smart Transportation SystemsIJCNCJournal
In -Vehicle Ad-Hoc Network (VANET), vehicles continuously transmit and receive spatiotemporal data with neighboring vehicles, thereby establishing a comprehensive 360-degree traffic awareness system. Vehicular Network safety applications facilitate the transmission of messages between vehicles that are near each other, at regular intervals, enhancing drivers' contextual understanding of the driving environment and significantly improving traffic safety. Privacy schemes in VANETs are vital to safeguard vehicles’ identities and their associated owners or drivers. Privacy schemes prevent unauthorized parties from linking the vehicle's communications to a specific real-world identity by employing techniques such as pseudonyms, randomization, or cryptographic protocols. Nevertheless, these communications frequently contain important vehicle information that malevolent groups could use to Monitor the vehicle over a long period. The acquisition of this shared data has the potential to facilitate the reconstruction of vehicle trajectories, thereby posing a potential risk to the privacy of the driver. Addressing the critical challenge of developing effective and scalable privacy-preserving protocols for communication in vehicle networks is of the highest priority. These protocols aim to reduce the transmission of confidential data while ensuring the required level of communication. This paper aims to propose an Advanced Privacy Vehicle Scheme (APV) that periodically changes pseudonyms to protect vehicle identities and improve privacy. The APV scheme utilizes a concept called the silent period, which involves changing the pseudonym of a vehicle periodically based on the tracking of neighboring vehicles. The pseudonym is a temporary identifier that vehicles use to communicate with each other in a VANET. By changing the pseudonym regularly, the APV scheme makes it difficult for unauthorized entities to link a vehicle's communications to its real-world identity. The proposed APV is compared to the SLOW, RSP, CAPS, and CPN techniques. The data indicates that the efficiency of APV is a better improvement in privacy metrics. It is evident that the AVP offers enhanced safety for vehicles during transportation in the smart city.
DEF: Deep Ensemble Neural Network Classifier for Android Malware DetectionIJCNCJournal
Malware is one of the threats to security of computer networks and information systems. Since malware instances are available sufficiently, there is increased interest among researchers on usage of Artificial Intelligence (AI). Of late AI-enabled methods such as machine learning (ML) and deep learning paved way for solving many real-world problems. As it is a learning-based approach, accumulated training samples help in improving thequality of training and thus leveraging malware detection accuracy. Existing deep learning methods are focusing on learning-based malware detection systems. However, there is need for improving the state of the art through ensemble approach. Towards this end, in this paper we proposed a framework known as Deep Ensemble Framework (DEF) for automatic malware detection. The framework obtains features from training samples. From given malware instance a grayscale image is generated. There is another process to extract the opcode sequences. Convolutional Neural Network (CNN) and Long Short Term Memory (LSTM) techniques are used to obtain grayscale image and opcode sequence respectively. Afterwards, a stacking ensemble is employed in order to achieve efficient malware detection and classification. Malware samples collected fromthe Internet sources and Microsoft are used for theempirical study. An algorithm known as Ensemble Learning for Automatic Malware Detection (EL-AML) is proposed to realize our framework. Another algorithm named Pre-Process is proposed to assist the EL-AML algorithm for obtaining intermediate features required by CNN and LSTM.Empirical study reveals that our framework outperforms many existing methods in terms of speed-up and accuracy.
High Performance NMF based Intrusion Detection System for Big Data IoT TrafficIJCNCJournal
With the emergence of smart devices and the Internet of Things (IoT), millions of users connected to the network produce massive network traffic datasets. These vast datasets of network traffic, Big Data are challenging to store, deal with and analyse using a single computer. In this paper we developed parallel implementation using a High Performance Computer (HPC) for the Non-Negative Matrix Factorization technique as an engine for an Intrusion Detection System (HPC-NMF-IDS). The large IoT traffic datasets of order of millions samples are distributed evenly on all the computing cores for both storage and speedup purpose. The distribution of computing tasks involved in the Matrix Factorization takes into account the reduction of the communication cost between the computing cores. The experiments we conducted on the proposed HPC-IDS-NMF give better results than the traditional ML-based intrusion detection systems. We could train the HPC model with datasets of one million samples in only 31 seconds instead of the 40 minutes using one processor), that is a speed up of 87 times. Moreover, we have got an excellent detection accuracy rate of 98% for KDD dataset.
IoT Guardian: A Novel Feature Discovery and Cooperative Game Theory Empowered...IJCNCJournal
Cyber intrusion attacks increasingly target the Internet of Things (IoT) ecosystem, exploiting vulnerable devices and networks. Malicious activities must be identified early to minimize damage and mitigate threats. Using actual benign and attack traffic from the CICIoT2023 dataset, this WORK aims to evaluate and benchmark machine-learning techniques for IoT intrusion detection. There are four main phases to the system. First, the CICIoT2023 dataset is refined to remove irrelevant features and clean up missing and duplicate data. The second phase employs statistical models and artificial intelligence to discover novel features. The most significant features are then selected in the third phase based on cooperative game theory. Using the original CICIoT2023 dataset and a dataset containing only novel features, we train and evaluate a variety of machine learning classifiers. On the original dataset, Random Forest achieved the highest accuracy of 99%. Still, with novel features, Random Forest's performance dropped only slightly (96%) while other models achieved significantly lower accuracy. As a whole, the work contributes substantial contributions to tailored feature engineering, feature selection, and rigorous benchmarking of IoT intrusion detection techniques. IoT networks and devices face continuously evolving threats, making it necessary to develop robust intrusion detection systems.
Enhancing Traffic Routing Inside a Network through IoT Technology & Network C...IJCNCJournal
IoT networking uses real items as stationary or mobile nodes. Mobile nodes complicate networking. Internet of Things (IoT) networks have a lot of control overhead messages because devices are mobile. These signals are generated by the constant flow of control data as such device identity, geographical positioning, node mobility, device configuration, and others. Network clustering is a popular overhead communication management method. Many cluster-based routing methods have been developed to address system restrictions. Node clustering based on the Internet of Things (IoT) protocol, may be used to cluster all network nodes according to predefined criteria. Each cluster will have a Smart Designated Node. SDN cluster management is efficient. Many intelligent nodes remain in the network. The network design spreads these signals. This paper presents an intelligent and responsive routing approach for clustered nodes in IoT networks. An existing method builds a new sub-area clustered topology. The Nodes Clustering Based on the Internet of Things (NCIoT) method improves message transmission between any two nodes. This will facilitate the secure and reliable interchange of healthcare data between professionals and patients. NCIoT is a system that organizes nodes in the Internet of Things (IoT) by grouping them together based on their proximity. It also picks SDN routes for these nodes. This approach involves selecting one option from a range of choices and preparing for likely outcomes problem addressing limitations on activities is a primary focus during the review process. Predictive inquiry employs the process of analyzing data to forecast and anticipate future events. This document provides an explanation of compact units. The Predictive Inquiry Small Packets (PISP) improved its backup system and partnered with SDN to establish a routing information table for each intelligent node, resulting in higher routing performance. Both principal and secondary roads are available for use. The simulation findings indicate that NCIoT algorithms outperform CBR protocols. Enhancements lead to a substantial 78% boost in network performance. In addition, the end-to-end latency dropped by 12.5%. The PISP methodology produces 5.9% more inquiry packets compared to alternative approaches. The algorithms are constructed and evaluated against academic ones.
IoT Guardian: A Novel Feature Discovery and Cooperative Game Theory Empowered...IJCNCJournal
Cyber intrusion attacks increasingly target the Internet of Things (IoT) ecosystem, exploiting vulnerable devices and networks. Malicious activities must be identified early to minimize damage and mitigate threats. Using actual benign and attack traffic from the CICIoT2023 dataset, this WORK aims to evaluate and benchmark machine-learning techniques for IoT intrusion detection. There are four main phases to the system. First, the CICIoT2023 dataset is refined to remove irrelevant features and clean up missing and duplicate data. The second phase employs statistical models and artificial intelligence to discover novel features. The most significant features are then selected in the third phase based on cooperative game theory. Using the original CICIoT2023 dataset and a dataset containing only novel features, we train and evaluate a variety of machine learning classifiers. On the original dataset, Random Forest achieved the highest accuracy of 99%. Still, with novel features, Random Forest's performance dropped only slightly (96%) while other models achieved significantly lower accuracy. As a whole, the work contributes substantial contributions to tailored feature engineering, feature selection, and rigorous benchmarking of IoT intrusion detection techniques. IoT networks and devices face continuously evolving threats, making it necessary to develop robust intrusion detection systems.
** Connect, Collaborate, And Innovate: IJCNC - Where Networking Futures Take ...IJCNCJournal
The International Journal of Computer Networks & Communications (IJCNC) is a bi monthly open access peer-reviewed journal that publishes articles which contribute new results in all areas of Computer Networks & Communications. The journal focuses on all technical and practical aspects of Computer Networks & data Communications. The goal of this journal is to bring together researchers and practitioners from academia and industry to focus on advanced networking concepts and establishing new collaborations in these areas.
Enhancing Traffic Routing Inside a Network through IoT Technology & Network C...IJCNCJournal
IoT networking uses real items as stationary or mobile nodes. Mobile nodes complicate networking. Internet of Things (IoT) networks have a lot of control overhead messages because devices are mobile. These signals are generated by the constant flow of control data as such device identity, geographical positioning, node mobility, device configuration, and others. Network clustering is a popular overhead communication management method. Many cluster-based routing methods have been developed to address system restrictions. Node clustering based on the Internet of Things (IoT) protocol, may be used to cluster all network nodes according to predefined criteria. Each cluster will have a Smart Designated Node. SDN cluster management is efficient. Many intelligent nodes remain in the network. The network design spreads these signals. This paper presents an intelligent and responsive routing approach for clustered nodes in IoT networks. An existing method builds a new sub-area clustered topology. The Nodes Clustering Based on the Internet of Things (NCIoT) method improves message transmission between any two nodes. This will facilitate the secure and reliable interchange of healthcare data between professionals and patients. NCIoT is a system that organizes nodes in the Internet of Things (IoT) by grouping them together based on their proximity. It also picks SDN routes for these nodes. This approach involves selecting one option from a range of choices and preparing for likely outcomes problem addressing limitations on activities is a primary focus during the review process. Predictive inquiry employs the process of analyzing data to forecast and anticipate future events. This document provides an explanation of compact units. The Predictive Inquiry Small Packets (PISP) improved its backup system and partnered with SDN to establish a routing information table for each intelligent node, resulting in higher routing performance. Both principal and secondary roads are available for use. The simulation findings indicate that NCIoT algorithms outperform CBR protocols. Enhancements lead to a substantial 78% boost in network performance. In addition, the end-to-end latency dropped by 12.5%. The PISP methodology produces 5.9% more inquiry packets compared to alternative approaches. The algorithms are constructed and evaluated against academic ones.
Enhancing Traffic Routing Inside a Network through IoT Technology & Network C...
OPTIMIZING VOIP USING A CROSS LAYER CALL ADMISSION CONTROL SCHEME
1. International Journal of Computer Networks & Communications (IJCNC) Vol.5, No.4, July 2013
DOI : 10.5121/ijcnc.2013.5410 117
OPTIMIZING VOIP USING A CROSS LAYER CALL
ADMISSION CONTROL SCHEME
Mumtaz AL-Mukhtar and Huda Abdulwahed
Department of Information Engineering, AL-Nahrain University, Baghdad, Iraq
almukhtar@fulbrightmail.org
ABSTRACT
Deploying wireless campus network becomes popular in many world universities for the services that are
provided. However, it suffers from different issues such as low VoIP network capacity, network congestion
effect on VoIP QoS and WLAN multi rate issue due to link adaptation technique. In this paper a cross layer
call admission control (CCAC) scheme is proposed to reduce the effects of these problems on VoWLAN
based on monitoring RTCPRR (Real Time Control Protocol Receiver Report) that provides the QoS level
for VoIP and monitoring the MAC layer for any change in the data rate. If the QoS level degrades due to
one of the aforementioned reasons, a considerable change in the packet size or the codec type will be the
solution. A wireless campus network is simulated using OPNET 14.5 modeler and many scenarios are
modeled to improve this proposed scheme.
KEYWORDS
VoIP Capacity, QoS, Cross-layering, VoWLAN, Codec Adaptation.
1. INTRODUCTION
Voice over IP (VoIP) has been widely used these years for its simplified infrastructure and
significant cost savings. One of the most interesting use cases for VoIP is in combination with the
IEEE 802.11 technology to provide wireless voice services to mobile devices such as laptops,
smart phones and PDAs. The use of VoWLAN (VoIP over WLAN) makes it possible for mobile
employee of an enterprise or a campus to be provided with cost effective voice and flexible
services [1]. However, two technical problems need to be solved. The first is that the system
capacity for voice can be quite low in WLAN. The second is that VoIP traffic and data traffic
from traditional applications (web, e-mail, etc...) can interfere with each other and bring down
VoIP performance.
Voice capacity is defined as the maximum number of voice sessions that can be supported
simultaneously by a network under specific quality constraints [2]. Therefore, it is essential to
determine the number of users a WLAN can support simultaneously without significantly
degrading the QoS and to analyze the delay, jitter and packet loss of VoIP over WLAN.
The objective of this paper is to study the effect of different codecs in IEEE 802.11 multi-rate
environment, and the VoIP packet payload size in order to develop a cross-layer call admission
control scheme between MAC and application layers. This mainly aims to enhance the network
capacity and quality for the VoIP calls with the accepted QoS constraints. This research focuses
on addressing the congestion in the network and the multi-rate issue caused by the link adaptation
technique in WLAN campus network.
The rest of the paper is organized as follows: section 2 introduces VoIP system, VoIP quality
evaluation criteria, and related link adaptation concept. Section 3 provides a brief overview of
several researches in the related area. Section 4 introduces the cross-layer call admission control
2. International Journal of Computer Networks & Communications (IJCNC) Vol.5, No.4, July 2013
118
scheme. The simulation scenarios and results are presented in section 5. Finally, conclusion is
drawn in section 6.
2. TECHNICAL BACKGROUND
This section gives a VoWLAN overview inspecting VoIP quality constraints. Our focus in this
research is to achieve better QoS under varying network conditions.
2.1. VoIP System
VoIP system consists of three essential components: codec, packetizer and playout buffer. At the
sender side, the analog voice signals are converted to digital signals, compressed and then
encoded by voice codecs. There are various voice codecs that are developed and standardized by
the International Telecommunication Union - Telecommunication Sector (ITU-T), such as G.711,
G.726, G.729, G.723.1a, etc. The subsequent process performed is packetization, where the
encoded voice is fragmented into equal size of packets by the packetizer. Each packet generated is
composed of the encoded voice and headers, which are added at different layers, specifically by
layers of Real-time Transport Protocol (RTP), User Datagram Protocol (UDP), and Internet
Protocol (IP), as well as by Data Link layer header. Moreover, RTP and Real-Time Control
Protocol (RTCP) are designed at the application layer to support real-time applications [3].
The packets are sent out over IP network to its destination, where the reverse process of decoding
and depacketizing of the received packets is carried out. Additionally, there are signaling
protocols of VoIP namely Session Initiation Protocol (SIP) and H.323. These protocols establish
VoIP calls. H.323 was standardized by ITU-T specifically to smoothly work together with PSTN.
On the other hand, SIP was standardized by Internet Engineering Task Force (IETF) to support
Internet applications, such as telephony. Figure 1 illustrates VoIP protocol stack with respect to
its TCP/IP protocol standard [4].
Figure 1. VoIP implementation in TCP/IP protocol standard
2.2. VoIP QoS Evaluation
The voice can be tested for quality in two ways, namely, subjective and objective. Humans
perform the subjective voice testing by listening to the voice sample, whereas, objective tests are
performed by computers. Common subjective benchmark for quantifying the performance of the
speech codec is the Mean Opinion Score (MOS). For performing MOS test, a voice sample is
given to a group of listeners. They listen to the sample and give a rating on a scale where
3. International Journal of Computer Networks & Communications (IJCNC) Vol.5, No.4, July 2013
119
"excellent" quality is given a score of 5, "good" a 4, "fair" a 3, "poor" a 2, and "'bad" a 1. The
ratings given by every member of the group is then averaged to get the MOS. The E-Model is
most commonly used for objective measurements. The basic result of the E-Model is the
calculation of the R-Factor. The R-factor is defined in terms of several parameters associated with
a voice channel across a mixed Switched Circuit Network and a Packet Switched Network. The
parameters included in the computation of the R factor are fairly extensive covering such factors
as echo, background noise, signal loss, codec impairments, and others. R-factor can be expressed
by (1) [5].
R = R0 + Is + Id + Ie + A (1)
Where, R0 groups the effects of noise, Is includes the effect of other impairments related to the
quantization of the voice signal, Id represents the impairment caused by delay, Ie covers the
impairments caused by low bit rate codecs and packet losses. The advantage factor A
compensates for the above impairments under various user conditions. For mobile telephony A is
assumed to be 10. We consider that A is 0 in the case of VoIP.
MOS is related to R-Factor by (2) [2].
For R<0: MOS=1
For R>100: MOS=4.5
For 0<R<100: MOS= 1+0.035R+7×10−6
R (R−60) (100−R) (2)
The voice performance is considered acceptable if the end-to-end delay is less than 150ms and the
packet loss rate is less than 2% [6].
2.3. IEEE 802.11 Standards and Link Adaptation
A variety of wireless LAN technologies exist that use various frequencies, modulation techniques,
and data rates. IEEE provides most of the standards, which are categorized as IEEE
802.11,802.11a,802.11b,802.11g,802.11e,…. ,and 802.11n. For instance 802.11b operates at 1, 2,
5.5, 11 Mbps, 802.11a & 802.11g can support up to 54 Mbps [7]. IEEE 802.11 supports the link
adaptation technique, which allows a wireless transmitter to select an appropriate transmission
rate on a packet basis according to the wireless link conditions. However, if a lower transmission
rate is selected because of deterioration of the wireless link conditions, packet delays and losses
will be increased due to the poor bandwidth availability, especially when the traffic load is high,
and as a result, the quality of real-time applications will significantly deteriorate. The IEEE
802.11 standard does not specify how and when to switch between the permitted rates, so various
link adaptation (LA) mechanisms have been studied [8]. One of the most effective solutions for
this issue is to adapt the codecs of some of the active voice flows using codec adaptation
algorithm (CAA) [9]. Cross layer methods have been designed to overcome numerous setbacks
that were previously faced by wireless applications running on the strict layered OSI protocol
stack. Cross layer design has numerous attributes. These include: fast stack adaptation, cross-
protocol coordination, ease of QoS provisioning to different applications, prioritization ease, no
unnecessary interaction between layers and improved throughput [10].
3. RELATED WORK
The two important issues related to VoWLAN are capacity and QoS depend on different
parameters such as data rate, speech codec, delay, jitter and packet loss. Several techniques and
algorithms have been presented to improve the quality of VoIP over WLAN. Cross-layer
4. International Journal of Computer Networks & Communications (IJCNC) Vol.5, No.4, July 2013
120
interaction approach has been introduced in several possible ways to optimize different layers’
parameters. Papapanagiotoua et al. [11] exploit the interrelations between data rate, packetization
interval, packet error rate, and retransmission attempts to produce a model to be used in the cross
layer call admission control scheme in order to optimize QoS for single cell WLAN.
Several researches [12, 13, 14, 15, 16] propose techniques to mitigate the effect of multi rate
issues on VoIP quality over WLAN by changing either packet size or codec type depending on
RTCP receiving messages and MAC layer data rate using either a cross layer control scheme or
an adaptive algorithm. Luthra and Sharma [17] suggest VoWLAN QoS enhancement using
numbers of QoS techniques like Integrated Services, Differentiated Services, and Resource
Reservation Protocol. While Chakraborty et al. [18] introduce QoS upgrading by optimizing the
concerned Access Points' parameters such as buffer size, retransmission limit, RTS threshold,
transmission power, antenna type, location factors and network load. In [19] IEEE 802.11e
standard has been harnessed to enhance the QoS compared with standard 802.11b/g.
The QoS for VoWLAN can be guaranteed using the smart call admission control scheme that is
based on a dynamic bandwidth channel allocation to reserve the greater BW for VoIP application
as introduced in [20]. However, authors in [21] and [22] invest for this purpose CAC (Call
Admission Control) based on channel load estimation that is built around the TBIT (Time
Between Idle Times) scheme. Jung et al. [23] employ call admission control scheme to achieve
high link utilization via investigation of the on-off patterns of VoIP traffic from the Brady model.
Chang et al. [24] propose a QoS – aware path switching strategy by using stream control
transmission protocol (SDCTP) in Multi–Protocol Label Switch (MPLS) network to improve
VoIP traffic. The main contribution of this paper is that, the proposed CCAC scheme will
enhance both the capacity and the QoS for VoWLAN in a campus-congested network based on
two parameters: packet size and codec type.
4. THE PROPOSED CROSS LAYER CALL ADMISSION CONTROL SCHEME
In order to improve the capacity and the QoS of VoIP over a WLAN, a cross layer call admission
control scheme (CCAC) is proposed to allow the communication between MAC and application
layers. This is depicted in figure 2.
Figure 2. CCAC proposed scheme
This scheme tries to increase network VoIP capacity and to reduce the effect of two problems: the
congestion and the WLAN link adaptation technique on VoIP calls. The periodic monitoring of
the RTCPRR (Real Time Control Protocol Receiving Report) messages will be helpful to find if
there is congestion in the network by evaluating the R-factor from the values of the end to end
delay and packet loss ratio parameters of these messages. If these values are greater than the
threshold values so that the end to end delay >150 ms and packet loss ratio >2% then the QoS for
5. International Journal of Computer Networks & Communications (IJCNC) Vol.5, No.4, July 2013
121
VoIP will be degraded indicating that R<70 which is minimum value for the accepted QoS. In
this case, the VoIP packet size will be checked whether it exceeds the predefined maximum size
for the negotiated codec. In negative case the packet size will be increased by one frame and the
monitoring process is resumed. However, if packet size exceeds the maximum then the codec will
be changed to the one of a lower bit rate. Table 1 illustrates parameters of different codecs. If the
transmission rate changes due to link adaptation algorithm then an alarm is send from the MAC
Layer to the CCAC scheme which also decides to either adjust the packet size or change the
codec type according to the RTCPRR. A descriptive flowchart of CCAC functions is shown in
figure 3.
Table 1. CODEC Parameters
CODEC Type G.711 G.726 G.729A G.723.1
Bit rate kbps 64 32 8 6.3
Bits per frame 8 4 80 159
Algorithmic delay (ms) 0.125 0.125 15 37.5
Codec delay (ms) 0.25 0.25 25 67.5
Compression type PCM ADPCM CSACELP ACELP
Complexity (MIPS) << 1 ≈ 1 <= 11 <=18
MOS 4.1 3.85 3.7 3.6
6. International Journal of Computer Networks & Communications (IJCNC) Vol.5, No.4, July 2013
122
Figure 3. CCAC Flow chart
5. SIMULATION & RESULTS
To assess how the proposed CCAC scheme can enhance the network performance in different
conditions, the university wireless campus network has been simulated using OPNET modeler
14.5 [25]. Many scenarios have been imposed using different packet size, data rate, and different
codecs to cover all possible cases.
5.1. Simulation Setup
The simulated wireless campus network consists of seven buildings; each one has three WLAN
subnets with two servers (FTP& DB) and a separate subnet of three wireless servers dedicated for
Email, Heavy browsing & Web searching applications. In addition there is an SIP sever for calls
management. This network is connected to the Internet as shown in figures 4, 5, and 6.
7. International Journal of Computer Networks & Communications (IJCNC) Vol.5, No.4, July 2013
123
Figure 4. Campus Network Figure 5. Building Subnet
Each WLAN subnet has 20 wireless work-station mobile nodes communicating with each other
through an Access Point. The nodes are using IEEE802.11b standard for their communication
with transmission rate of 11 Mbps. Six applications are defined in this network which are VoIP,
FTP, DB, Email, web searching, and heavy browsing. The codec chosen for VoIP is G.711 with
one frame per packet. The VoIP call is assumed to be between two nodes, caller and the callee.
Figure 6. Server's Subnet
5.2. Simulation Scenarios and Results
Simulation scenarios cover two main cases as follows:
5.2.1. Effects of Congestion and Addition of New VoIP User
The first scenario aims to find how many simultaneous calls can be achieved in the network with
packet size of one frame and G.711codec for VoIP application. Only five calls can be found in the
same time so that the required end to end delay, MOS, and packet loss will be acceptable. Adding
new calls or other traffic cause the performance to be unacceptable. These results are depicted in
8. International Journal of Computer Networks & Communications (IJCNC) Vol.5, No.4, July 2013
124
figures 7 and 8. In the second scenario, the VoIP packet size has been increased to comprise two
frames. In this scenario, 10 simultaneous VoIP calls can be found and other traffic could be
added. In the third scenario, the packet size has been increased to 3 frames to achieve 14 calls
with more added traffic. In the fourth scenario the number of frames (F) considered is 4 and the
simultaneous calls (N) are 18. In the fifth scenario F=5 and N=64 with each node in the network
has at least three running application so that the network is congested. The MOS is found to be
>3, end to end delay <150 s and packet loss ratio <2% which are accepted. Adding more frames
to the VoIP packet of the G.711 codec will produce unacceptable results. Figures 9a& b show the
maximum number of N and MOS values with respect to F. Next scenario is considered when
there is a new call, a codec change for this call to G.726 of 32 Kbps with 2 frames per packet will
occur for this call to be accepted.
Figure 7. End to End Delay (case1, scenario1)
Figure 8. Traffic send and Traffic received (case1, scenario1)
9. International Journal of Computer Networks & Communications (IJCNC) Vol.5, No.4, July 2013
125
Figure 9a. Max. N with respect to F Figure 9b. MOS value with respect to F
5.2.2. Effect of Multi Rate Issue
Many scenarios have been introduced to cover changes in transmission rate from 11 to 5.5, 2, and
1 Mbps for certain VoIP nodes. The packet size has been varied from 1 to 5 frames and the codec
deployed is G.711. In the first scenario, the VoIP packet contains one frame and the transmission
rate varies to 5.5 Mbps. Only the end-to-end delay will be affected but it is still accepted.
Nevertheless, when a rate of 2 & 1 Mbps is deployed, the performance will be unacceptable. By
increasing the packet size to 2 frames the performance is enhanced as shown in figures 10 and 11.
Figure 10. End to End Delay (case 2, scenario 2)
Figure 11. MOS value (case 2, scenario 1)
10. International Journal of Computer Networks & Communications (IJCNC) Vol.5, No.4, July 2013
126
The second scenario is considered when the VoIP node’s transmission rate is 11Mbps but the
packet size is 2 frames and a transmission rate changes within 5.5, 2, and 1 Mbps. A change in the
packet size is made to enhance the VoIP quality of service parameters so it can be accepted. This
is shown in figures 12 & 13. When the transmission rate is 2 Mbps and the frame size is 2 frames
the end to end delay is not acceptable i.e. > 150 ms. However, at a packet size of 3 frames the end
to end delay will be acceptable.
Figure 12. End to End Delay (case 2, scenario2)
Figure 13. MOS value (case 2, scenario 2)
The third scenario is considered when VoIP packet size is 3 frames at transmission rate of 11
Mbps and it varies to 5.5, 2, and 1Mbps. At 5.5& 2 Mbps, the packet size changes to 4 frames to
get the accepted QoS parameters values. At 1 Mbps increasing the packet size will not enhance
the performance so the codec type will be switched to G.726 with packet size of 4 frames as
shown in figures 14&15.
11. International Journal of Computer Networks & Communications (IJCNC) Vol.5, No.4, July 2013
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Figure 14. End to End delay (case 2, scenario 3)
Figure 15. MOS value (case 2, scenario 3)
The fourth scenario is considered when the VoIP packet size consists of 4 frames and some VoIP
nodes change their transmission rate from 11 Mbps to 5.5, 2, 1 Mbps. At 5.5Mbps changing the
codec will be more effective than packet size so G.726 will be the codec. At 2 Mbps also the
codec will be changed to the one of a lower bit rate which is G.729 A. That is even with 5 frames
packet size of the G.726 codec the quality would not be acceptable. At 1Mbps G.729A with 5
VoIP frames also does not provide good quality so it is replaced with G.723.1 of 6.3 Kbps so the
calls could be continued. These results are depicted in figures 16&17.
12. International Journal of Computer Networks & Communications (IJCNC) Vol.5, No.4, July 2013
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Figure 16. End to End delay (case 2, scenario 4)
Figure 17. MOS value (case 2, scenario 4)
6. CONCLUSION
In this paper, the VoWLAN application is tested for performance in campus network. A cross
layer call admission control scheme is proposed to enhance network capacity and QoS for VoIP
by adapting packet size or codec type. The simulation results proved that the proposed CCAC
scheme provides an efficient and fast way for solving the network congestion effect on VoIP
application. Increasing network VoIP capacity and reducing the multi rate effect is achieved by
monitoring the RTCPRR for the last end to end delay and Packet loss ratio that provides the
network QoS level, which is represented by R-factor. As well as the MAC layer is monitored for
any changes in the data rate of the mobile node. If the QoS degrades due to one of the above
problems then the solution will be by either varying the packet size or codec type to optimize
network performance without dropping the call.
13. International Journal of Computer Networks & Communications (IJCNC) Vol.5, No.4, July 2013
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REFERENCES
[1] X. Wei, Y. Bouslimani, & K. Sellal, (2012) “VoIP Based Solution for the Use over a Campus
Environment” 25th IEEE Canadian Conference on Electrical and Computer Engineering
(CCECE), PP 1-5.
[2] C. Brouzioutis , V. Vitsas & P. Chatzimisios, (2010) “Studying the Impact of Data Traffic on
Voice Capacity in IEEE 802.11 WLANs”, IEEE International Conference on Communications, PP
1-6.
[3] M. N. Ismail, (2010) “Analysis of Secure Real Time Transport Protocol on VoIP over Wireless
LAN in Campus Environment” International Journal on Computer Science and Engineering
(IJCSE), Volume 02, No. 03, PP 898-902.
[4] S. A. A. Alshakhsi & H. Hasbullah, (2010) “Improving QoS of VoWLAN via Cross-Layer
Interaction Approach”, International Symposium on Information Technology, PP 678-682.
[5] A. Mukhopadhyay, T. Chakraborty, S. Bhunia, I. Misra & S. Sanyal, (2011) “Study of Enhanced
VoIP Performance under Congested Wireless Network Scenarios”, Third International Conference
on Communication Systems and Networks (COMSNETS 2011), PP 1-7.
[6] W. Wang, S. Liew, & V. Li, (2005) “Solutions to Performance Problems in VoIP Over a 802.11
Wireless LAN”, IEEE Transactions on Vehicular Technology journal, Volume 54, Issue 1, PP
366-384.
[7] D. E. COMER, (2008) Computer Networks and Internets, Pearson Education, Inc., Fifth Edition.
[8] T. Kawata, & H. Yamada, (2006) “Adaptive Multi-Rate VoIP for IEEE 802.11 Wireless Networks
with Link Adaptation Function”, Proceedings of the Global Telecommunications Conference
(GLOBECOM '06), PP 1-5.
[9] Z. Chen, L. Wang, F. Zhang, X. Wang & W. Chen, (2008) “VoIP over WLANs by Adapting
Transmitting Interval and Call Admission Control”, IEEE International Conference on
Communications, PP 3242-3246.
[10] V. Sentongo & H. Chan, (2009) “Optimization of Quality of Service Requirements for Real-Time
Applications using Cross Layer Design”, IEEE AFRICON'09, PP 1-9.
[11] I. Papapanagiotoua, F. Granellib, D. Kliazovichb, & M. Devetsikiotis, (2011) “A Metamodeling
Approach for Cross-Layer Optimization: A Framework and Application to Voice over WiFi”,
Simulation Modelling Practice and Theory Journal, Volume 19, Issue 9, PP 2117-2129.
[12] S. Ali Alshakhsi, & H. Hasbullah, (2011) “Improving QoS of VoWLAN via Cross-Layer-Based
Adaptive Approach”, International Conference on Information Science and Applications, PP 1-8.
[13] M. Tuysus & H. Mantar, (2010) “Evaluation of Cross Layer QoS Approach for Improving Voice
Quality over Multi Rate WLANs”, International Conference on Computer Engineering and
Systems (ICCES), PP 73-78.
[14] A. Sfairopoulou, B. Bellalta, C. Macian, & C. Oliver, (2011) “A Comparative Survey of Adaptive
Codec Solutions for VoIP over Multirate WLANs: A Capacity versus Quality Performance Trade-
Off” EURASIP Journal on Wireless Communications and Networking, Volume 2011, Issue 1, PP
1-13.
[15] H. Kazemitabar & A. Saida, (2011) “An Adaptive Rate Control Algorithm for VoIP over Multi-
Rate WLANs”, 2nd World Conference on Information Technology (WCIT-2011), PP 1087-1092.
[16] P. McGovern, P. Perry & S. Murphy, (2011) “Endpoint-Based Call Admission Control and
Resource Management for VoWLAN”, IEEE Transactions on Mobile Computing, Volume 10,
NO. 5, PP 684-699.
[17] P. Luthra & M. Sharma, (2012) “Performance Evaluation of Audio Codecs using VoIP Traffic in
Wireless LAN using RSVP”, International Journal of Computer Applications, Volume 40, No.7,
PP 15-21.
[18] T. Chakraborty, A. Mukhopadhyay, S. Bhunia, I. S. Misra & S. K. Sanyal, (2012) “An
Optimization Technique for Improved VoIP Performance over Wireless LAN”, Journal of
Networks, Volume 7, NO. 3, PP 480-493.
[19] S. V. Bhanu & R. M. Chandrasekaran, (2012) “Voice Call Capacity Analysis and Enhancement of
IEEE 802.11 WLAN”, European Journal of Scientific Research, Volume 76, No.2, PP 271-280.
[20] R. Ganiga, B. Muniyal & Pradeep, (2012) “ Characteristic Analysis of VoIP Traffic for Wireless
Networks In Comparison with CBR using QualNet Network Simulator” International Journal of
Computer Applications, Volume 50, No.11, PP 25-31.
14. International Journal of Computer Networks & Communications (IJCNC) Vol.5, No.4, July 2013
130
[21] K. Yasukawa, A. G. Forte & H. Schulzrinne, (2007) “Distributed Delay Estimation and Call
Admission Control in IEEE 802.11 WLANs”, IEEE International Conference on Network
Protocols, PP 334-335.
[22] P. Dini, N. Baldo & J. Nin-Guerrero, (2010) “Distributed Call Admission Control for VoIP over
802.11 WLANs based on Channel Load Estimation” IEEE International Conference on
Communications journal, PP 1-6.
[23] Jung Ji-Young , Seo Dong-Yoon & Lee Jung-Ryun, (2013) "VoIP Call Admission Control
Scheme Considering Voip on-off Patterns", International Conference on Information Networking
(ICOIN), PP 371 – 374.
[24] Chang Lin-huang, Lee Tsung-Han, Chu Hung-Chi, Lo Yu-Lung & Chen Yu-Jen, (2013) "QoS-
Aware Path Switching for VoIP Traffic using SCTP", Computer Standards & Interfaces Journal,
Volume 35, Issue 1, PP 158–169.
[25] A. S. Sethi & V.Y. Hnatyshin, (2013) The Practical OPNET User Guide for Computer Network
Simulation, CRC Press.