At the last few years, multimedia communication has
been developed and improved rapidly in order to
enable users to communicate between each other over
the internet. Generally, multimedia communication
consists of audio and video communication. However,
this research concentrates on audio conferencing
only. The audio translation between protocols is a
very critical issue, because it solves the communic
ation
problems between any two protocols. So, it enables
people around the world to talk with each other eve
n
they use different protocols. In this research, a r
eal time audio translation module between two proto
cols
has been done. These two protocols are: InterAsteri
sk eXchange Protocol (IAX) and Real-Time Switching
Control Protocol (RSW), which they are widely used
to provide two ways audio transfer feature. The
solution here is to provide interworking between th
e two protocols which they have different media
transports, audio codec’s, header formats and diffe
rent transport protocols for the audio transmission
. This
translation will help bridging the gap between the
two protocols by providing interworking capability
between the two audio streams of IAX and RSW. Some
related works have been done to provide translation
between IAX and RSW control signalling messages. Bu
t, this research paper concentrates on the
translation that depends on the media transfer. The
proposed translation module was tested and evaluat
ed
in different scenarios in order to examine its perf
ormance. The obtained results showed that the Real-
Time
Audio Translation Module produces lower rates of pa
cket delay and jitter than the acceptance values fo
r
each of the mentioned performance metrics.
IP resides at the network layer and provides logical addressing that allows systems on different logical networks to communicate. It is a connectionless protocol that does not provide reliability, flow control, or sequencing. VoIP uses RTP, which sits atop UDP, to transport real-time voice data in an efficient manner without retransmissions. UDP is used instead of TCP for VoIP as reliability is less important than latency for real-time voice communications.
IP resides at the network layer of the OSI model and provides logical addressing that allows systems on different logical networks to communicate. IP packets contain source and destination addresses as well as other fields. Transport protocols like UDP and TCP run on top of IP, with UDP being connectionless and used for real-time voice traffic in VoIP due to its simplicity and lower latency compared to TCP, which provides reliability but higher latency through mechanisms like acknowledgments and retransmissions. RTP runs on top of UDP to provide additional timestamping and sequencing information important for applications like voice calling.
Transport Layer Port or TCP/IP & UDP PortNetwax Lab
A port is an application-specific or process-specific software construct serving as a communications
endpoint in a computer's host operating system. The purpose of ports is to uniquely identify different
applications or processes running on a single computer and thereby enable them to share a single
physical connection to a packet-switched network like the Internet. In the context of the Internet
Protocol, a port is associated with an IP address of the host, as well as the type of protocol used for
communication.
This document provides an overview of IP and how it enables Voice over IP (VoIP). It discusses the OSI model and how IP fits as the network layer. It describes IP packet fields and how transport protocols like UDP and TCP work with IP. UDP is often used for real-time audio in VoIP instead of TCP due to its lower latency. RTP adds sequencing and timing information on top of UDP for VoIP.
This document summarizes the study of parameters that determine the quality of service of various Voice over IP (VoIP) clients. The study measured parameters like bandwidth requirement, delay, packet size and observed how clients behaved under different network conditions. Key findings were that bandwidth, jitter, latency and packet loss most affected quality of service. The VoIP clients tested included Google Talk, Skype, VQube, Windows Live Messenger and Yahoo Voice Messenger. Network Address Translation (NAT) types and Simple Traversal of UDP through NAT (STUN) were also explained.
The document discusses MPLS (Multi-Protocol Label Switching) including traditional IP forwarding, IP over ATM, MPLS concepts, MPLS architecture, MPLS forwarding, MPLS applications, MPLS protocols, and forwarding equivalence classes. MPLS combines the advantages of connection-oriented forwarding with IP routing by assigning labels to packets and forwarding based on those labels rather than long IP addresses.
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.
This document provides an overview of LTE networks and technology. It discusses key aspects of LTE including peak data rates of 50-100 Mbps, reduced latency under 10ms, OFDMA for downlink and SC-FDMA for uplink, support for bandwidths from 1.4-20 MHz, and mobility support up to 350km/h. It also examines the architecture including elements such as the eNodeB, MME, S-GW, P-GW, and interfaces such as S1, X2.
IP resides at the network layer and provides logical addressing that allows systems on different logical networks to communicate. It is a connectionless protocol that does not provide reliability, flow control, or sequencing. VoIP uses RTP, which sits atop UDP, to transport real-time voice data in an efficient manner without retransmissions. UDP is used instead of TCP for VoIP as reliability is less important than latency for real-time voice communications.
IP resides at the network layer of the OSI model and provides logical addressing that allows systems on different logical networks to communicate. IP packets contain source and destination addresses as well as other fields. Transport protocols like UDP and TCP run on top of IP, with UDP being connectionless and used for real-time voice traffic in VoIP due to its simplicity and lower latency compared to TCP, which provides reliability but higher latency through mechanisms like acknowledgments and retransmissions. RTP runs on top of UDP to provide additional timestamping and sequencing information important for applications like voice calling.
Transport Layer Port or TCP/IP & UDP PortNetwax Lab
A port is an application-specific or process-specific software construct serving as a communications
endpoint in a computer's host operating system. The purpose of ports is to uniquely identify different
applications or processes running on a single computer and thereby enable them to share a single
physical connection to a packet-switched network like the Internet. In the context of the Internet
Protocol, a port is associated with an IP address of the host, as well as the type of protocol used for
communication.
This document provides an overview of IP and how it enables Voice over IP (VoIP). It discusses the OSI model and how IP fits as the network layer. It describes IP packet fields and how transport protocols like UDP and TCP work with IP. UDP is often used for real-time audio in VoIP instead of TCP due to its lower latency. RTP adds sequencing and timing information on top of UDP for VoIP.
This document summarizes the study of parameters that determine the quality of service of various Voice over IP (VoIP) clients. The study measured parameters like bandwidth requirement, delay, packet size and observed how clients behaved under different network conditions. Key findings were that bandwidth, jitter, latency and packet loss most affected quality of service. The VoIP clients tested included Google Talk, Skype, VQube, Windows Live Messenger and Yahoo Voice Messenger. Network Address Translation (NAT) types and Simple Traversal of UDP through NAT (STUN) were also explained.
The document discusses MPLS (Multi-Protocol Label Switching) including traditional IP forwarding, IP over ATM, MPLS concepts, MPLS architecture, MPLS forwarding, MPLS applications, MPLS protocols, and forwarding equivalence classes. MPLS combines the advantages of connection-oriented forwarding with IP routing by assigning labels to packets and forwarding based on those labels rather than long IP addresses.
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.
This document provides an overview of LTE networks and technology. It discusses key aspects of LTE including peak data rates of 50-100 Mbps, reduced latency under 10ms, OFDMA for downlink and SC-FDMA for uplink, support for bandwidths from 1.4-20 MHz, and mobility support up to 350km/h. It also examines the architecture including elements such as the eNodeB, MME, S-GW, P-GW, and interfaces such as S1, X2.
Comparative study of various voip applications in 802.11 a wireless network s...ijmnct
Today, Voice over Wireless Local Area Network (VOWLAN) is the most accepted Internet application.
There are a large number of literatures regarding the performance of various WLAN networks. Most of
them focus on simulations and modeling, but there are also some experiments with real networks. This
paper explains the comparison of performance of two different VOIP (Voice over Internet Protocol)
applications over the same IEEE 802.11a wireless network. Radio link standard 802.11a have maximum
transmission rate of 54Mbps. First protocol is session initiation protocol (SIP) and second is H.323
protocol. First one has an agent called SIP proxy. Second have a gateway reflects the characteristics of a
Switched Circuit Network (SCN). With this comparison we have required to obtain a better understanding
of wireless network suitability for voice communication in IP network.
Dar es Salaam institute of Technology (DIT) provides training on digital networks including 3G and 4G mobile technologies. 3G networks introduced higher speed packet data and mobile multimedia services compared to previous generations. UMTS/WCDMA is an IMT-2000 3G standard that supports voice and fast packet data through technologies like HSDPA and HSUPA which enable peak downlink rates of 14.4 Mbps and uplink rates of 5.8 Mbps. HSPA+ further increases speeds through MIMO and higher order modulations.
This document provides an overview of Asynchronous Transfer Mode (ATM) technology. It discusses:
- What ATM is and why it was developed to provide high-speed, low delay networking for various traffic types like voice, video, and data.
- Key aspects of ATM including fixed-length 53-byte cells, virtual connections, connection-oriented and connectionless modes, and quality of service guarantees.
- Components of the ATM protocol stack including the physical layer, ATM layer, and ATM adaptation layer (AAL). It describes the different AAL types.
- ATM network architecture including interfaces like UNI and NNI and the use of virtual paths and channels for
VoIP uses packet networks to carry voice calls in addition to data. It works by converting analog voice signals to digital data packets which are transmitted over IP networks and reconverted to analog at the receiving end. Key components include IP phones, signaling servers, and protocols like SIP and H.323 which handle call setup and signaling. Quality of service for VoIP depends on factors like packet loss, delay, and jitter which can be managed through queuing and reserving bandwidth for voice traffic.
The document discusses the transport layer in computer networks. It provides 3 key points:
1. The transport layer provides a service to the application layer and obtains a service from the network layer. It is responsible for multiplexing/demultiplexing, reliable data transfer, flow control, and congestion control.
2. The main transport layer protocols used in the Internet are UDP (connectionless) and TCP (connection-oriented). TCP provides reliable in-order byte streams with congestion control and flow control. UDP is unreliable and unordered.
3. Reliable data transfer protocols like RDT use mechanisms like checksums, acknowledgements, negative acknowledgements, retransmissions, and sequence numbers to
The document discusses SIP (Session Initiation Protocol) mobility support over wireless networks. It provides an overview of SIP, including its components and message types. It then describes the SIP handoff procedure, wherein the mobile host sends a JOIN message to initiate soft handoff between base stations. Finally, it analyzes sources of handoff delay in the SIP mobility management process and finds that the major portion of delay is due to the wireless link. Delays of over 6 seconds are estimated, which can negatively impact real-time media streams.
This document provides an overview of MPLS (Multi-Protocol Label Switching) including:
1) It describes the need for MPLS arising from limitations in traditional IP forwarding and issues running one statmux technology over another.
2) It explains basic MPLS concepts like label switching, label distribution protocols, label edge and switch routers, label switching paths, and forwarding equivalence classes.
3) It outlines the basic working process of MPLS including label encapsulation, lookup, and processing functions like push, pop and swap.
The document discusses the application layer in computer networking. It describes the client-server model where clients send queries to servers which respond with answers. It also discusses name resolution, where hostnames are translated to IP addresses, and protocols like TCP and UDP which provide transport services. Common applications like email, the web, and peer-to-peer are briefly mentioned as examples.
IMS is an IP-based architecture that enables the delivery of multimedia services over both fixed and wireless networks. It provides a common service delivery platform for various access networks and allows for convergence of services. Key benefits of IMS include enabling a user-centric network, reducing costs through network resource sharing, and providing a consistent user experience across multiple devices and access networks through a single user identity and profile.
This document summarizes key concepts about the transport layer in computer networks. It discusses:
1. The transport layer is responsible for process-to-process delivery of data across a network. This involves delivering packets from one process to another, often using a client-server model.
2. There are two main transport layer protocols - UDP, which is a connectionless and unreliable protocol, and TCP, which establishes connections and provides reliable data delivery.
3. TCP and UDP use port numbers along with IP addresses to uniquely identify processes. TCP also implements flow and error control to ensure reliable data transfer.
This document provides an overview of Integrated Services Digital Network (ISDN). It describes ISDN as a set of standards that define end-to-end digital connectivity for carrying voice, data, and videos concurrently. The key benefits of ISDN include higher speeds, faster call setup times, and the ability to run voice and data simultaneously. ISDN services include Basic Rate Interface (BRI) and Primary Rate Interface (PRI). BRI provides 2 B channels for user data and 1 D channel for control, while PRI provides more B channels and higher speeds for carrier use. The document also outlines ISDN devices, reference points, call processing, and its relationship to the OSI model layers.
Wireless local area networks (WLANs) connect devices within a local area using radio waves instead of wires. They became popular in the 1980s-1990s as a way to share resources like printers and storage over a local area network (LAN). Almost all modern WLANs use the 802.11 standard and operate in the 2.4GHz or 5GHz spectrum. A typical WLAN consists of an access point that connects wireless clients like laptops and smartphones to a wired network backbone. WLANs provide flexibility but come with challenges around interference and security.
The document discusses transport layer protocols and services including:
- TCP provides reliable, in-order delivery through congestion control, flow control, and connection setup. UDP provides unreliable, unordered delivery with no connection.
- Transport protocols multiplex and demultiplex data between applications using port numbers. TCP uses a 4-tuple of IP addresses and port numbers to identify each connection.
- UDP is useful for streaming multimedia since it is loss tolerant but rate sensitive, while TCP provides reliability through congestion control and retransmissions.
This document discusses local area network (LAN) technologies, with a focus on Ethernet. It outlines the following objectives:
- Briefly discuss dominant wired LANs including Ethernet and other media types.
- Describe Media Access Control (MAC) and Carrier Sense Multiple Access with Collision Detection (CSMA/CD).
- Explain the Address Resolution Protocol (ARP) and bridges.
- Discuss switched Ethernet and virtual LANs (VLANs).
The document then provides details on Ethernet frames, MAC addresses, CSMA/CD, cabling standards and specifications.
presentation on TCP/IP protocols data comunicationsAnyapuPranav
The document provides an overview of the TCP/IP protocol architecture. It discusses the five layers of TCP/IP including the physical, network access, internet, transport, and application layers. It describes the protocols used at each layer, such as IP, TCP, UDP, HTTP, and FTP. The document also discusses how data is encapsulated as it passes through each layer of the TCP/IP model and is transmitted from one host to another across networks and the internet.
The document summarizes key topics related to transport layer protocols:
- It describes the services provided by the transport layer, including addressing, connection establishment and release, flow control, and multiplexing.
- It provides details on common transport protocols like TCP and UDP, including their packet headers, connection management, congestion control, and performance issues at high speeds.
- It also presents an example transport protocol and uses finite state machines to model its operation and connection management.
This document provides an overview of a course on broadband and TCP/IP fundamentals. It discusses the topics that will be covered in each of the four sessions, including basics of TCP/IP networks, switching and scheduling, routing and transport, and applications and security. It also lists some recommended textbooks and references for the course.
Bluetooth is a wireless technology standard that allows short-range connections between devices like mobile phones, headphones, and laptops using radio waves in the 2.4 GHz spectrum. It uses frequency hopping spread spectrum technology and establishes piconets between one master device and up to seven slave devices to enable communication between connected devices. Bluetooth supports both synchronous and asynchronous connections and can be used to transfer data, voice, and interface with other wireless protocols like TCP/IP.
IMPORTANCE OF REALISTIC MOBILITY MODELS FOR VANET NETWORK SIMULATIONIJCNCJournal
In the performance evaluation of a protocol for a vehicular ad hoc network, the protocol should be tested under a realistic conditions including, representative data traffic models, and realistic movements of the mobile nodes which are the vehicles (i.e., a mobility model). This work is a comparative study between two mobility models that are used in the simulations of vehicular networks, i.e., MOVE (MObility model generator for VEhicular networks) and CityMob, a mobility pattern generator for VANET. We describe several mobility models for VANET simulations.
In this paper we aim to show that the mobility models can significantly affect the simulation results in VANET networks. The results presented in this article prove the importance of choosing a suitable real world scenario for performances studies of routing protocols in this kind of network.
Corporate role in protecting consumers from the risk of identity theftIJCNCJournal
The Internet has made it possible for users to be robbed of their reputation, money and credit worthiness by
the click of a mouse. The impact of identity theft severely limits victims’ ability to participate in commerce,
education and normal societal functions. This paper evaluates resurgence in syndicated cyber attacks,
which includes but not limited to identity theft, corporate espionage and cyber warfare taking advantage of
the Internet as a medium of operations. The paper highlights the increase of cyber related attacks in the
past ten years due to lack of transatlantic international corporation between participating countries,
coherent information security policies, data aggregation and sound international laws to facilitate
prosecution of perpetrators. The cyber space coupled with availability of free hacking tools has contributed
to resurgence in syndicated identity theft, corporate espionage and identity theft by organized crime
elements taking advantage of the Internet as a medium of operations. This paper presents conclusive
solution that users, organizations and consumers can enact to protect themselves from the threat of cyber
attacks culminating into identity theft, financial loss or both.
Comparative study of various voip applications in 802.11 a wireless network s...ijmnct
Today, Voice over Wireless Local Area Network (VOWLAN) is the most accepted Internet application.
There are a large number of literatures regarding the performance of various WLAN networks. Most of
them focus on simulations and modeling, but there are also some experiments with real networks. This
paper explains the comparison of performance of two different VOIP (Voice over Internet Protocol)
applications over the same IEEE 802.11a wireless network. Radio link standard 802.11a have maximum
transmission rate of 54Mbps. First protocol is session initiation protocol (SIP) and second is H.323
protocol. First one has an agent called SIP proxy. Second have a gateway reflects the characteristics of a
Switched Circuit Network (SCN). With this comparison we have required to obtain a better understanding
of wireless network suitability for voice communication in IP network.
Dar es Salaam institute of Technology (DIT) provides training on digital networks including 3G and 4G mobile technologies. 3G networks introduced higher speed packet data and mobile multimedia services compared to previous generations. UMTS/WCDMA is an IMT-2000 3G standard that supports voice and fast packet data through technologies like HSDPA and HSUPA which enable peak downlink rates of 14.4 Mbps and uplink rates of 5.8 Mbps. HSPA+ further increases speeds through MIMO and higher order modulations.
This document provides an overview of Asynchronous Transfer Mode (ATM) technology. It discusses:
- What ATM is and why it was developed to provide high-speed, low delay networking for various traffic types like voice, video, and data.
- Key aspects of ATM including fixed-length 53-byte cells, virtual connections, connection-oriented and connectionless modes, and quality of service guarantees.
- Components of the ATM protocol stack including the physical layer, ATM layer, and ATM adaptation layer (AAL). It describes the different AAL types.
- ATM network architecture including interfaces like UNI and NNI and the use of virtual paths and channels for
VoIP uses packet networks to carry voice calls in addition to data. It works by converting analog voice signals to digital data packets which are transmitted over IP networks and reconverted to analog at the receiving end. Key components include IP phones, signaling servers, and protocols like SIP and H.323 which handle call setup and signaling. Quality of service for VoIP depends on factors like packet loss, delay, and jitter which can be managed through queuing and reserving bandwidth for voice traffic.
The document discusses the transport layer in computer networks. It provides 3 key points:
1. The transport layer provides a service to the application layer and obtains a service from the network layer. It is responsible for multiplexing/demultiplexing, reliable data transfer, flow control, and congestion control.
2. The main transport layer protocols used in the Internet are UDP (connectionless) and TCP (connection-oriented). TCP provides reliable in-order byte streams with congestion control and flow control. UDP is unreliable and unordered.
3. Reliable data transfer protocols like RDT use mechanisms like checksums, acknowledgements, negative acknowledgements, retransmissions, and sequence numbers to
The document discusses SIP (Session Initiation Protocol) mobility support over wireless networks. It provides an overview of SIP, including its components and message types. It then describes the SIP handoff procedure, wherein the mobile host sends a JOIN message to initiate soft handoff between base stations. Finally, it analyzes sources of handoff delay in the SIP mobility management process and finds that the major portion of delay is due to the wireless link. Delays of over 6 seconds are estimated, which can negatively impact real-time media streams.
This document provides an overview of MPLS (Multi-Protocol Label Switching) including:
1) It describes the need for MPLS arising from limitations in traditional IP forwarding and issues running one statmux technology over another.
2) It explains basic MPLS concepts like label switching, label distribution protocols, label edge and switch routers, label switching paths, and forwarding equivalence classes.
3) It outlines the basic working process of MPLS including label encapsulation, lookup, and processing functions like push, pop and swap.
The document discusses the application layer in computer networking. It describes the client-server model where clients send queries to servers which respond with answers. It also discusses name resolution, where hostnames are translated to IP addresses, and protocols like TCP and UDP which provide transport services. Common applications like email, the web, and peer-to-peer are briefly mentioned as examples.
IMS is an IP-based architecture that enables the delivery of multimedia services over both fixed and wireless networks. It provides a common service delivery platform for various access networks and allows for convergence of services. Key benefits of IMS include enabling a user-centric network, reducing costs through network resource sharing, and providing a consistent user experience across multiple devices and access networks through a single user identity and profile.
This document summarizes key concepts about the transport layer in computer networks. It discusses:
1. The transport layer is responsible for process-to-process delivery of data across a network. This involves delivering packets from one process to another, often using a client-server model.
2. There are two main transport layer protocols - UDP, which is a connectionless and unreliable protocol, and TCP, which establishes connections and provides reliable data delivery.
3. TCP and UDP use port numbers along with IP addresses to uniquely identify processes. TCP also implements flow and error control to ensure reliable data transfer.
This document provides an overview of Integrated Services Digital Network (ISDN). It describes ISDN as a set of standards that define end-to-end digital connectivity for carrying voice, data, and videos concurrently. The key benefits of ISDN include higher speeds, faster call setup times, and the ability to run voice and data simultaneously. ISDN services include Basic Rate Interface (BRI) and Primary Rate Interface (PRI). BRI provides 2 B channels for user data and 1 D channel for control, while PRI provides more B channels and higher speeds for carrier use. The document also outlines ISDN devices, reference points, call processing, and its relationship to the OSI model layers.
Wireless local area networks (WLANs) connect devices within a local area using radio waves instead of wires. They became popular in the 1980s-1990s as a way to share resources like printers and storage over a local area network (LAN). Almost all modern WLANs use the 802.11 standard and operate in the 2.4GHz or 5GHz spectrum. A typical WLAN consists of an access point that connects wireless clients like laptops and smartphones to a wired network backbone. WLANs provide flexibility but come with challenges around interference and security.
The document discusses transport layer protocols and services including:
- TCP provides reliable, in-order delivery through congestion control, flow control, and connection setup. UDP provides unreliable, unordered delivery with no connection.
- Transport protocols multiplex and demultiplex data between applications using port numbers. TCP uses a 4-tuple of IP addresses and port numbers to identify each connection.
- UDP is useful for streaming multimedia since it is loss tolerant but rate sensitive, while TCP provides reliability through congestion control and retransmissions.
This document discusses local area network (LAN) technologies, with a focus on Ethernet. It outlines the following objectives:
- Briefly discuss dominant wired LANs including Ethernet and other media types.
- Describe Media Access Control (MAC) and Carrier Sense Multiple Access with Collision Detection (CSMA/CD).
- Explain the Address Resolution Protocol (ARP) and bridges.
- Discuss switched Ethernet and virtual LANs (VLANs).
The document then provides details on Ethernet frames, MAC addresses, CSMA/CD, cabling standards and specifications.
presentation on TCP/IP protocols data comunicationsAnyapuPranav
The document provides an overview of the TCP/IP protocol architecture. It discusses the five layers of TCP/IP including the physical, network access, internet, transport, and application layers. It describes the protocols used at each layer, such as IP, TCP, UDP, HTTP, and FTP. The document also discusses how data is encapsulated as it passes through each layer of the TCP/IP model and is transmitted from one host to another across networks and the internet.
The document summarizes key topics related to transport layer protocols:
- It describes the services provided by the transport layer, including addressing, connection establishment and release, flow control, and multiplexing.
- It provides details on common transport protocols like TCP and UDP, including their packet headers, connection management, congestion control, and performance issues at high speeds.
- It also presents an example transport protocol and uses finite state machines to model its operation and connection management.
This document provides an overview of a course on broadband and TCP/IP fundamentals. It discusses the topics that will be covered in each of the four sessions, including basics of TCP/IP networks, switching and scheduling, routing and transport, and applications and security. It also lists some recommended textbooks and references for the course.
Bluetooth is a wireless technology standard that allows short-range connections between devices like mobile phones, headphones, and laptops using radio waves in the 2.4 GHz spectrum. It uses frequency hopping spread spectrum technology and establishes piconets between one master device and up to seven slave devices to enable communication between connected devices. Bluetooth supports both synchronous and asynchronous connections and can be used to transfer data, voice, and interface with other wireless protocols like TCP/IP.
IMPORTANCE OF REALISTIC MOBILITY MODELS FOR VANET NETWORK SIMULATIONIJCNCJournal
In the performance evaluation of a protocol for a vehicular ad hoc network, the protocol should be tested under a realistic conditions including, representative data traffic models, and realistic movements of the mobile nodes which are the vehicles (i.e., a mobility model). This work is a comparative study between two mobility models that are used in the simulations of vehicular networks, i.e., MOVE (MObility model generator for VEhicular networks) and CityMob, a mobility pattern generator for VANET. We describe several mobility models for VANET simulations.
In this paper we aim to show that the mobility models can significantly affect the simulation results in VANET networks. The results presented in this article prove the importance of choosing a suitable real world scenario for performances studies of routing protocols in this kind of network.
Corporate role in protecting consumers from the risk of identity theftIJCNCJournal
The Internet has made it possible for users to be robbed of their reputation, money and credit worthiness by
the click of a mouse. The impact of identity theft severely limits victims’ ability to participate in commerce,
education and normal societal functions. This paper evaluates resurgence in syndicated cyber attacks,
which includes but not limited to identity theft, corporate espionage and cyber warfare taking advantage of
the Internet as a medium of operations. The paper highlights the increase of cyber related attacks in the
past ten years due to lack of transatlantic international corporation between participating countries,
coherent information security policies, data aggregation and sound international laws to facilitate
prosecution of perpetrators. The cyber space coupled with availability of free hacking tools has contributed
to resurgence in syndicated identity theft, corporate espionage and identity theft by organized crime
elements taking advantage of the Internet as a medium of operations. This paper presents conclusive
solution that users, organizations and consumers can enact to protect themselves from the threat of cyber
attacks culminating into identity theft, financial loss or both.
PERFORMANCE EVALUATION OF THE EFFECT OF NOISE POWER JAMMER ON THE MOBILE BLUE...IJCNCJournal
This document evaluates the effect of noise power jamming on Bluetooth personal area networks (PANs). It finds that barrage noise jamming across the full 79MHz Bluetooth band is ineffective from 10 meters away. Narrower 20MHz and 5MHz sweep jamming can reduce the processing gain over time by causing channels to be blocked by the adaptive frequency hopping (AFH) mechanism. Jamming power levels of 2-5 watts were still insufficient to overcome path loss and processing gain at distances over 1 meter. Future work could evaluate follower jamming techniques targeting the frequency hopping scheme.
ANALYSIS OF IPV6 TRANSITION TECHNOLOGIESIJCNCJournal
Currently IPv6 is extremely popular with companies, organizations and Internet service providers (ISP)
due to the limitations of IPv4. In order to prevent an abrupt change from IPv4 to IPv6, three mechanisms
will be used to provide a smooth transition from IPv4 to IPv6 with minimum effect on the network. These
mechanisms are Dual-Stack, Tunnel and Translation. This research will shed the light on IPv4 and IPv6
and assess the automatic and manual transition strategies of the IPv6 by comparing their performances in
order to show how the transition strategy affects network behaviour. The experiment will be executed using
OPNET Modeler that simulates a network containing a Wide Area Network (WAN) , a Local Area Network
(LAN), hosts and servers. The results will be presented in graphs and tables, with further explanation. The
experiment will use different measurements such as throughput, latency (delay), queuing delay, and TCP
delay.
On modeling controller switch interaction in openflow based sdnsIJCNCJournal
With an increase in number of software defined network (SDN) deployments,and OpenFlow consolidating as the protocol of choice for controller-switch interactions, a need to analytically model the system for performance analysis is increasing. An attempt has previously been made in [1] to model the syste considering both a controller and a switch as an M/M/1 queue. The method, although useful, lacks accuracy for higher probabilities of new flows entering the network. The approach is also deficient of
details on how it can be extended to more than one node in the data plane.These two short-comings are addressed in this paper where thecontroller and switch are modeled
collectively as Jackson’s network, with essential tuning to suit OpenFlow-based SDN. The consequent analysis shows the resilience of the model even for higher number of new flow entries. An example is also used
to illustrate the case of multiple nodes in the data plane.
A bandwidth allocation model provisioning framework with autonomic characteri...IJCNCJournal
The Bandwidth Allocation Models (MAM, RDM, G-RDM and AllocTC-Sharing) are management
alternatives currently available which propose different resource (bandwidth) allocation strategies in
multiservice networks. The BAM adoption by a network is typically a management choice and
configuration task executed by the network operations and management system setup in a static or nearly
static way. This paper proposes and explores the alternative of allowing BAM definition and configuration
on a more dynamic way. In effect, one of the basic motivations towards BAM dynamic allocation is the fact
that multiservice networks characteristics (traffic load) may change considerably in daily network
operation and, as such, some dynamics in BAM allocation should be introduced in order to improve
performance. A framework is presented supporting BAM dynamic allocation. The framework adopts an
OpenFlow-based software-defined networking (SDN) implementation approach in order to support
scalability issues with a centralized controller and management network view. The framework architecture
also supports the implementation of some autonomic characteristics which, in brief, look for improving and
facilitating the decision-making process involved with BAM provisioning in a multiservice network. A
proof of concept is presented evaluating different BAM performance under different traffic loads in order to
demonstrate the framework strategy adopted.
On client’s interactive behaviour to design peer selection policies for bitto...IJCNCJournal
Peer-to-peer swarming protocols have been proven to be very efficient for content replication over Internet.
This fact has certainly motivated proposals to adapt these protocols to meet the requirements of on-demand
streaming system. The vast majority of these proposals focus on modifying the piece and peer selection
policies, respectively, of the original protocols. Nonetheless, it is true that more attention has often been
given to the piece selection policy rather than to the peer selection policy. Within this context, this article
proposes a simple algorithm to be used as basis for peer selection policies of BitTorrent-like protocols,
considering interactive scenarios. To this end, we analyze the client’s interactive behaviour when accessing
real multimedia systems. This analysis consists of looking into workloads of real content providers and
assessing three important metrics, namely temporal dispersion, spatial dispersion and object position
popularity. These metrics are then used as the main guidelines for writing the algorithm. To the best of our
knowledge, this is the first time that the client’s interactive behaviour is specially considered to derive an
algorithm for peer selection policies. Finally, the conclusion of this article is drawn with key challenges
and possible future work in this research field.
Link aware nice application level multicast protocolIJCNCJournal
Multicast is one of the most efficient ways to dist
ribute data to multiple users. There are different
types of
Multicast such as IP Multicast, Overlay Multicast,
and Application Layer Multicast (ALM). In this pape
r,
we present a link-aware Application Layer (ALM) Mul
ticast algorithm. Our proposed algorithm, Link
Aware-NICE (LA-NICE) [1], is an enhanced version of
the NICE protocol [2]. LA-NICE protocol uses the
variations of bandwidth or capacity in communicatio
n links to improve multicast message delivery and
minimize end-to-end delay. OMNeT++ simulation frame
work [3] was used to evaluate LA-NICE. The
evaluation is done through a comparison between LA-
NICE and NICE. The simulation results showed that
LA-NICE produces an increased percentage of success
ful message delivery ranging from 2% to 10%
compared to NICE. Also, LA-NICE has less average de
lay and less average message hop count than NICE
which reduces the overall latency of message delive
ry
A new method for controlling and maintainingIJCNCJournal
Topology Control is an essential technique in a wireless sensor network to extend the operational time of
the sensor nodes. The goal of this technique is to maintain network connectivity and optimize performance
metrics such as network lifetime and throughput. In this paper we presented a new method for controlling
and maintaining topology in wireless sensor networks that show some improvement over the state of art
methods. The results are analyzed based on objective criteria.
Energy aware clustering protocol (eacp)IJCNCJournal
The document summarizes an Energy Aware Clustering Protocol (EACP) proposed for heterogeneous wireless sensor networks. EACP introduces heterogeneity by using two types of nodes: normal and advanced. Normal nodes elect cluster heads using a probability scheme based on residual and average energy. Advanced nodes use a separate probability scheme and act as gateways for normal cluster heads, transmitting their data to the base station. The performance of EACP is compared to SEP through simulations, showing better results for stability period, network life and energy savings.
Design, implementation and evaluation of icmp based available network bandwid...IJCNCJournal
We propose a method to measure available network ba
ndwidth using the Internet Control Message
Protocol (ICMP). The recently proposed ImTCP techni
que uses Transmission Control Protocol (TCP) data
packets and the corresponding acknowledgement respo
nses to measure the available bandwidth between
sender and receiver. Since ImTCP needs to change th
e sender’s TCP implementation, it needs
modifications to sender’s operating system kernel.
Moreover, ImTCP cannot measure available bandwidth
accurately if the receiver sends delayed acknowledg
ments. These problems stem from the use of TCP. In
this paper, we discuss an ICMP-based method that ov
ercomes these limitations. We evaluate the
performance of the proposed method in an experiment
al network and show that it generates less
measurement traffic and requires less time for band
width measurement than PathLoad. We also show that
proposed method can measure the available bandwidth
even if the bandwidth changes during
measurement
Security analysis of generalized confidentialmodulation for quantum communica...IJCNCJournal
We propose a new evaluation method for‘generalized confidential modulation(GCM)’ for quantum
communication. Confidential modulationrealizes a secret communication by using secret information for
modulationand noise in a channel. Y-00 is one of the famous methods of GCM forquantum communication.
The existing evaluation methods for GCM arebased on stream ciphers. They can estimate its analytical
security andthe evaluation depends on the security status of pseudo random numbergenerator (PRNG)
which controls the modulation. On the other hand,our method is based on mode of operation for block
ciphers and clears theweaknesses from structural viewpoint. Using our method, we can comparethe
security of different GCM structures. Our method of security evaluationand comparison does not depend on
the security status of PRNG.From the results of our evaluation, we conclude that the security of GCMis
limited to computational security.
In recent years, cooperative communication is a hot topic of research and it is a powerful physical layer
technique to combat fading in wireless relaying scenario. Concerning with the physical layer issues, in this
paper it is focussed on with providing a better space time block coding (STBC) scheme and incorporating it
in the cooperative relaying nodes to upgrade the system performance. Recently, the golden codes have
proven to exhibit a superior performance in a wireless MIMO (Multiple Input Multiple Output) scenario
than any other code. However, a serious limitation associated with it is its increased decoding complexity.
This paper attempts to resolve this challenge through suitable modification of golden code such that a less
complex sphere decoder could be used without much compromising the error rates. The decoder complexity
is analyzed through simulation and it proves to exhibit less complexity compared to the conventional
(Maximum likelihood) ML decoder. The single relay cooperative STBC consisting of source, relay and
destination are considered. The cooperative protocol strategy considered in the relay node is Decode and
forward (DF) protocol. The proposed modified golden code with less complex sphere decoder is
implemented in the nodes of the cooperative relaying system to achieve better performance in the system.
The simulation results have validated the effectiveness of the proposed scheme by offering better BER
performance, minimum outage probability and increased spectral efficiency compared to the non
cooperative transmission method.
Due to the proliferation in the number of users that are accessing the internet and due to the increase in
the number of the electronic devices that support mobility like mobiles, laptops and many others that
definitely lead to the need of a protocol that supports a mobility. Mobile Internet Protocol is a
recommended Internet protocol designed to support the mobility of a user (host). This protocol provides a
continuous connectivity for any mobile host . In the traditional Mobile IP all packets forwarded to the
Mobile host from the correspondent node will be forwarded via the Home Agent (HA) and that leads to
the triangle routing problem .
(ISP MBG) technique is used as a route optimization technique for solving the triangle routing problem
in conventional Mobile IPv4. This technique has been implemented on .net platform .The study of this
technique was discussed before using 2 similar Internet Service Providers and the simulation results
provided a better performance compared with the Conventional Mobile IP Technique. In this paper the
simulator will be used to study the performance of the (ISP MBG) technique using two different Internet
Service Providers ( ISPs) structures separated by a single Mobile Border Gateway ( MBG).Simulation
results shows also a better performance compared with the conventional Mobile IP technique .
Towards internet of things iots integration of wireless sensor network to clo...IJCNCJournal
Cloud computing provides great benefits for applications hosted on the Web that also have special
computational and storage requirements. This paper proposes an extensible and flexible architecture for
integrating Wireless Sensor Networks with the Cloud. We have used REST based Web services as an
interoperable application layer that can be directly integrated into other application domains for remote
monitoring such as e-health care services, smart homes, or even vehicular area networks (VAN). For proof
of concept, we have implemented a REST based Web services on an IP based low power WSN test bed,
which enables data access from anywhere. The alert feature has also been implemented to notify users via
email or tweets for monitoring data when they exceed values and events of interest.
A comparative analysis of number portability routing schemesIJCNCJournal
To reap the benefits of liberalized telecom market, the implementation of number portability (NP) is utmost
important. NP allows end user to retain their telephone number in case of change of geographical location
or service type or service provider. This paper describes the various number portability routing schemes
namely, All Call Query, Query on Release, Call Dropback and Onward routing. The comparative analysis
between these routing schemes on various parameters is presented here. The issues pertaining to NP have
also been described.
Pwm technique to overcome the effect ofIJCNCJournal
Many current communication systems suffer from performance degradation due to the high sensitivity to
high power peaks especially in the nonlinear devices. The author introduces a new concept based on the
Pulse Width Modulation (PWM), namely MIMO-OFDM system based PWM (MO-PWM) to overcome this
deficiency. Here, the peak-to-average power ratio (PAPR) problem in Orthogonal Frequency Division
Multiplexing (OFDM) technique is used as a criterion to check the validity of the proposed work.
Moreover, the proposed system work has been implemented over Field Programmable Gate Array (FPGA),
which is designed to characterize both of the complexity and the speed issues.
The systems performance based MO-PWM and validity have been checked based on a numerical analysis
and a conducted simulation. The simulation results show that the MO-PWM can clearly reduce the PAPR
values nevertheless the used OFDM systems’ specifications, and gives a promising results over some
techniques found in the literature, such as clipping, SLM and PTS under same bandwidth occupancy and
system’s specifications.
Cloud computing challenges and solutionsIJCNCJournal
Cloud computing is an emerging area of computer technology that benefits form the processing power and
the computing resources of many connected, geographically distanced computers connected via Internet.
Cloud computing eliminates the need of having a complete infrastructure of hardware and software to meet
users requirements and applications. It can be thought of or considered as a complete or a partial
outsourcing of hardware and software resources. To access cloud applications, a good Internet connection
and a standard Internet browser are required. Cloud computing has its own drawback from the security
point of view; this paper aims to address most of these threats and their possible solutions.
Correlation based feature selection (cfs) technique to predict student perfro...IJCNCJournal
Education data mining is an emerging stream which h
elps in mining academic data for solving various
types of problems. One of the problems is the selec
tion of a proper academic track. The admission of a
student in engineering college depends on many fact
ors. In this paper we have tried to implement a
classification technique to assist students in pred
icting their success in admission in an engineering
stream.We have analyzed the data set containing inf
ormation about student’s academic as well as socio-
demographic variables, with attributes such as fami
ly pressure, interest, gender, XII marks and CET ra
nk
in entrance examinations and historical data of pre
vious batch of students. Feature selection is a pro
cess
for removing irrelevant and redundant features whic
h will help improve the predictive accuracy of
classifiers. In this paper first we have used featu
re selection attribute algorithms Chi-square.InfoGa
in, and
GainRatio to predict the relevant features. Then we
have applied fast correlation base filter on given
features. Later classification is done using NBTree
, MultilayerPerceptron, NaiveBayes and Instance bas
ed
–K- nearest neighbor. Results showed reduction in c
omputational cost and time and increase in predicti
ve
accuracy for the student model
The document discusses technologies for optimizing voice quality over wireless networks. It describes Voice over IP (VoIP) and challenges in providing quality of service (QoS) over wireless local area networks (WLANs). It then introduces Samsung's Voice-aware Traffic Scheduling (VaTS) and Network Controlled Voice Optimization (NCVO) algorithms, which aim to improve QoS for voice traffic on enterprise WLANs. VaTS reduces overhead by combining voice packets for different clients into a single transmission. NCVO finely tunes MAC parameters like contention window sizes according to network conditions to prioritize voice traffic.
This document discusses throughput performance analysis of Voice over IP (VoIP) in Long Term Evolution (LTE) networks. It begins with an introduction to LTE and the increasing demand for high-speed wireless communication. It then describes the generic frame structures used in LTE, including Type 1 and Type 2 frames for Frequency Division Duplexing (FDD) and Time Division Duplexing (TDD) respectively. Next, it covers LTE's quality of service framework and use of Real-time Transport Protocol (RTP) for audio and video transmission. Finally, it provides an overview of VoIP technology and its characteristics, such as delay requirements and use of codecs like AMR to provide constant bit rate transmission of compressed
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.
This section provides background on VoIP applications and their components. It discusses how VoIP works by transmitting voice packets over IP networks, unlike traditional PSTN which uses circuit switching. The two main VoIP protocols are H.323 and SIP, which provide call control and setup. UDP is typically used instead of TCP for transmitting VoIP due to its lower overhead and lack of error checking, which is tolerable for voice transmission. IPv4 and IPv6 can both be used to transmit VoIP, with IPv6 having a larger header size.
Optimal Communication Of Real Time Data On Secure Cdma Ip...Stefanie Yang
The document discusses optimal communication of real-time data over secure CDMA IP RAN networks. It proposes a new coding approach for security enhancement and quality improvement based on spectrum utilization and antenna coding. A traffic model is developed for transmitting image and audio data over wireless channels with AWGN noise and fading effects. An optimal spectrum sensing approach is developed for resource allocation and communicated using the secure WeP protocol. Results show improvement in quality metrics for the proposed system compared to conventional CDMA modeling.
OPTIMIZING VOIP USING A CROSS LAYER CALL ADMISSION CONTROL SCHEMEIJCNCJournal
This document discusses optimizing VoIP quality over wireless networks using a cross-layer call admission control scheme. It proposes monitoring real-time control protocol reports and data rates at the MAC layer to determine when quality is degraded. When quality degrades due to issues like network congestion or variable transmission rates, the solution is to adapt the packet size or codec type. The proposed scheme is simulated using a wireless campus network model to improve performance.
The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications including WebRTC, television services and web-based push-to-talk features.
2. A Survey of Rate Adaptation Techniques for Dynamic Adaptive Streaming over...AliIssa53
This document provides a survey of rate adaptation techniques for Dynamic Adaptive Streaming over HTTP (DASH). It discusses the evolution of video delivery over IP networks, including early use of UDP/RTP and development of standards like DASH. Rate adaptation is important for DASH to adjust video quality based on changing network conditions. The document categorizes rate adaptation techniques according to the feedback signals used and whether adaptation is done at the client, server, or network. It also reviews studies on measuring video traffic.
Analysis of VoIP Traffic in WiMAX EnvironmentEditor IJMTER
This document reviews several studies that analyzed the performance of VoIP traffic over WiMAX networks using different VoIP codecs and WiMAX service classes. It summarizes the findings of various papers on how QoS parameters like throughput, delay, jitter compared for codecs like G.711, G.723, G.729 when using the UGS, rtPS, nrtPS and BE service classes. Most studies found that UGS generally performed best for VoIP due to its ability to guarantee bandwidth and minimize jitter and delay, while G.711 typically provided the best voice quality. The document aims to compare the results across different service classes and codecs.
Video steaming Throughput Performance Analysis over LTEiosrjce
This document analyzes the video streaming throughput performance over LTE networks using the OPNET simulation tool. It simulates two scenarios: 1) downlink and uplink video conferencing with static users and 2) the same with users moving at 30m/s. The key metrics measured are packet delay variation and end-to-end delay. The results show that static users experience higher packet delay variation than mobile users, likely due to increased traffic accumulation. End-to-end delay is also higher for static users compared to those moving at 30m/s.
This document analyzes the video streaming throughput performance over LTE networks using the OPNET simulation tool. It simulates two scenarios: 1) downlink and uplink video conferencing with static users and 2) the same with users moving at 30m/s. The key metrics measured are packet delay variation and end-to-end delay. The results show that static users experience higher packet delay variation than mobile users, likely due to increased traffic accumulation. End-to-end delay is also higher for static users compared to those moving at 30m/s.
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.
Mobile Networking and Ad hoc routing protocols validationIOSR Journals
This document discusses mobile networking and ad hoc routing protocols. It begins with an overview of cellular phone networks and their growth in usage. It then describes mobile ad hoc networks and some of the challenges in designing routing protocols for them. The document evaluates two model checking tools, SPIN and UPPAAL, and discusses their ability to verify properties of ad hoc routing protocols through formal validation methods.
dynamic media streaming over wireless and ip networksNaveen Dubey
The document discusses internet and wireless technologies such as IEEE 802.11, HTTP, and Mobile IP. It describes experiments on dynamic media streaming over wireless networks using different transport protocols like TCP and PRRT. The experiments showed that TCP suffers from throughput variations on wireless networks leading to underutilization of bandwidth. In contrast, PRRT, which implements predictably reliable transport, was able to optimize bandwidth utilization for media streaming over wireless and mobile internet paths.
This document discusses a video aware QoS MAC protocol for wireless video sensor networks. It begins with background on wireless multimedia sensor networks and challenges in transmitting video over such networks with limited bandwidth. It then discusses the IEEE 802.11e standard, which defines four access categories (ACs) with different priorities to provide quality of service. However, 802.11e has static resource allocation that does not adapt to network conditions. The proposed VAQMAC protocol allows dynamic allocation of transmission opportunity (TXOP) limits and changing frame priorities during congestion under a modified 802.11e to better support video transmission over wireless sensor networks.
The document discusses media VoIP protocols and technologies. It provides an overview of:
- RTP and RTCP protocols for transporting audio and video over IP networks. RTP provides sequencing and time stamping while RTCP provides quality feedback.
- Common audio and video codecs like G.711, H.261, H.263 that are used to compress media.
- Mechanisms for mixing and translating streams to connect networks of different capabilities.
- DTMF signaling which can be carried in-band or out-of-band using SIP INFO.
ON THE SUPPORT OF MULTIMEDIA APPLICATIONS OVER WIRELESS MESH NETWORKS ijwmn
This document summarizes research on supporting multimedia applications over wireless mesh networks. It proposes an efficient routing algorithm and a QoS approach at the MAC layer. The routing algorithm aims to transport multimedia traffic with QoS requirements. The MAC layer approach improves 802.11e to better facilitate video transport over the mesh network. Related work on routing protocols, QoS solutions at the network and MAC layers, and clustering approaches are also reviewed.
This document summarizes a study on the performance of real-time and non-real-time traffic in IEEE 802.11 wireless local area networks (WLANs) using the network simulator NS2. The study evaluates the impact of the distributed coordination function (DCF) on throughput, packet loss, and delay. It describes simulations with various traffic types, including voice, video, and data, under different load conditions. The results show the packet loss, throughput, and delay for each simulation case.
WAVE (Wireless Access in Vehicular Environments) uses dedicated short-range communications (DSRC) technology to enable high-speed vehicle-to-vehicle and vehicle-to-infrastructure communication. It has applications in intelligent transportation systems for vehicle safety and Internet access. A WAVE system consists of roadside units (RSUs) that broadcast information to onboard units (OBUs) in passing vehicles. OBUs listen for safety information on a default control channel and can join local wireless networks called WAVE basic service sets to receive additional information on other channels.
The document discusses Synchronous Optical Networking (SONET) and Synchronous Digital Hierarchy (SDH), which are standardized protocols for transmitting multiple digital signals over fiber optic cables. They were developed to replace older asynchronous systems and allow synchronized transport of data from different sources. Key features include high transmission rates up to 40Gbps, simple addition and removal of low-rate channels, high reliability through automatic backup mechanisms, and future compatibility with new services. The main differences between SONET and SDH are their standardized bit rates which were chosen to integrate existing network technologies.
Similar to Real time audio translation module between iax and rsw (20)
Particle Swarm Optimization–Long Short-Term Memory based Channel Estimation w...IJCNCJournal
Paper Title
Particle Swarm Optimization–Long Short-Term Memory based Channel Estimation with Hybrid Beam Forming Power Transfer in WSN-IoT Applications
Authors
Reginald Jude Sixtus J and Tamilarasi Muthu, Puducherry Technological University, India
Abstract
Non-Orthogonal Multiple Access (NOMA) helps to overcome various difficulties in future technology wireless communications. NOMA, when utilized with millimeter wave multiple-input multiple-output (MIMO) systems, channel estimation becomes extremely difficult. For reaping the benefits of the NOMA and mm-Wave combination, effective channel estimation is required. In this paper, we propose an enhanced particle swarm optimization based long short-term memory estimator network (PSOLSTMEstNet), which is a neural network model that can be employed to forecast the bandwidth required in the mm-Wave MIMO network. The prime advantage of the LSTM is that it has the capability of dynamically adapting to the functioning pattern of fluctuating channel state. The LSTM stage with adaptive coding and modulation enhances the BER.PSO algorithm is employed to optimize input weights of LSTM network. The modified algorithm splits the power by channel condition of every single user. Participants will be first sorted into distinct groups depending upon respective channel conditions, using a hybrid beamforming approach. The network characteristics are fine-estimated using PSO-LSTMEstNet after a rough approximation of channels parameters derived from the received data.
Keywords
Signal to Noise Ratio (SNR), Bit Error Rate (BER), mm-Wave, MIMO, NOMA, deep learning, optimization.
Volume URL: https://airccse.org/journal/ijc2022.html
Abstract URL:https://aircconline.com/abstract/ijcnc/v14n5/14522cnc05.html
Pdf URL: https://aircconline.com/ijcnc/V14N5/14522cnc05.pdf
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#adhocnetwork #VANETs #OLSRrouting #routing #MPR #nderesidualenergy #korea #cognitiveradionetworks #radionetworks #rendezvoussequence
Here's where you can reach us : ijcnc@airccse.org or ijcnc@aircconline.com
June 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.
Enhanced Traffic Congestion Management with Fog Computing - A Simulation-Base...IJCNCJournal
Abstract: Accurate latency computation is essential for the Internet of Things (IoT) since the connected
devices generate a vast amount of data that is processed on cloud infrastructure. However, the cloud is not
an optimal solution. To overcome this issue, fog computing is used to enable processing at the edge while
still allowing communication with the cloud. Many applications rely on fog computing, including traffic
management. In this paper, an Intelligent Traffic Congestion Mitigation System (ITCMS) is proposed to
address traffic congestion in heavily populated smart cities. The proposed system is implemented using fog
computing and tested in a crowdedCairo city. The results obtained indicate that the execution time of the
simulation is 4,538 seconds, and the delay in the application loop is 49.67 seconds. The paper addresses
various issues, including CPU usage, heap memory usage, throughput, and the total average delay, which
are essential for evaluating the performance of the ITCMS. Our system model is also compared with other
models to assess its performance. A comparison is made using two parameters, namely throughput and the
total average delay, between the ITCMS, IOV (Internet of Vehicle), and STL (Seasonal-Trend
Decomposition Procedure based on LOESS). Consequently, the results confirm that the proposed system
outperforms the others in terms of higher accuracy, lower latency, and improved traffic efficiency.
Call for Papers -International Journal of Computer Networks & Communications ...IJCNCJournal
International Journal of Computer Networks & Communications (IJCNC)
Citations, h-index, i10-index of IJCNC
---- Scopus, ERA Listed, WJCI Indexed ----
Scopus Cite Score 2022--1.8
https://airccse.org/journal/ijcnc.html
IJCNC is listed in ERA 2023 as per the Australian Research Council (ARC) Journal Ranking
Scope & Topics
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.
Authors are solicited to contribute to this journal by submitting articles that illustrate research results, projects, surveying works and industrial experiences that describe significant advances in the Computer Networks & Communications.
Topics of Interest
· Network Protocols & Wireless Networks
· Network Architectures
· High speed networks
· Routing, switching and addressing techniques
· Next Generation Internet
· Next Generation Web Architectures
· Network Operations & management
· Adhoc and sensor networks
· Internet and Web applications
· Ubiquitous networks
· Mobile networks & Wireless LAN
· Wireless Multimedia systems
· Wireless communications
· Heterogeneous wireless networks
· Measurement & Performance Analysis
· Peer to peer and overlay networks
· QoS and Resource Management
· Network Based applications
· Network Security
· Self-Organizing Networks and Networked Systems
· Optical Networking
· Mobile & Broadband Wireless Internet
· Recent trends & Developments in Computer Networks
Paper Submission
Authors are invited to submit papers for this journal through E-mail: ijcnc@airccse.org or through Submission System. Submissions must be original and should not have been published previously or be under consideration for publication while being evaluated for this Journal.
Important Dates
· Submission Deadline : June 22, 2024
· Notification : July 22, 2024
· Final Manuscript Due : July 29, 2024
· Publication Date : Determined by the Editor-in-Chief
Contact Us
Here's where you can reach us: ijcnc@airccse.org or ijcnc@aircconline.com
For other details please visit - http://airccse.org/journal/ijcnc.html
Rendezvous Sequence Generation Algorithm for Cognitive Radio Networks in Post...IJCNCJournal
Recent natural disasters have inflicted tremendous damage on humanity, with their scale progressively increasing and leading to numerous casualties. Events such as earthquakes can trigger secondary disasters, such as tsunamis, further complicating the situation by destroying communication infrastructures. This destruction impedes the dissemination of information about secondary disasters and complicates post-disaster rescue efforts. Consequently, there is an urgent demand for technologies capable of substituting for these destroyed communication infrastructures. This paper proposes a technique for generating rendezvous sequences to swiftly reconnect communication infrastructures in post-disaster scenarios. We compare the time required for rendezvous using the proposed technique against existing methods and analyze the average time taken to establish links with the rendezvous technique, discussing its significance. This research presents a novel approach enabling rapid recovery of destroyed communication infrastructures in disaster environments through Cognitive Radio Network (CRN) technology, showcasing the potential to significantly improve disaster response and recovery efforts. The proposed method reduces the time for the rendezvous compared to existing methods, suggesting that it can enhance the efficiency of rescue operations in post-disaster scenarios and contribute to life-saving efforts.
Blockchain Enforced Attribute based Access Control with ZKP for Healthcare Se...IJCNCJournal
The relationship between doctors and patients is reinforced through the expanded communication channels provided by remote healthcare services, resulting in heightened patient satisfaction and loyalty. Nonetheless, the growth of these services is hampered by security and privacy challenges they confront. Additionally, patient electronic health records (EHR) information is dispersed across multiple hospitals in different formats, undermining data sovereignty. It allows any service to assert authority over their EHR, effectively controlling its usage. This paper proposes a blockchain enforced attribute-based access control in healthcare service. To enhance the privacy and data-sovereignty, the proposed system employs attribute-based access control, zero-knowledge proof (ZKP) and blockchain. The role of data within our system is pivotal in defining attributes. These attributes, in turn, form the fundamental basis for access control criteria. Blockchain is used to keep hospital information in public chain but EHR related data in private chain. Furthermore, EHR provides access control by using the attributed based cryptosystem before they are stored in the blockchain. Analysis shows that the proposed system provides data sovereignty with privacy provision based on the attributed based access control.
EECRPSID: Energy-Efficient Cluster-Based Routing Protocol with a Secure Intru...IJCNCJournal
A revolutionary idea that has gained significance in technology for Internet of Things (IoT) networks backed by WSNs is the " Energy-Efficient Cluster-Based Routing Protocol with a Secure Intrusion Detection" (EECRPSID). A WSN-powered IoT infrastructure's hardware foundation is hardware with autonomous sensing capabilities. The significant features of the proposed technology are intelligent environment sensing, independent data collection, and information transfer to connected devices. However, hardware flaws and issues with energy consumption may be to blame for device failures in WSN-assisted IoT networks. This can potentially obstruct the transfer of data. A reliable route significantly reduces data retransmissions, which reduces traffic and conserves energy. The sensor hardware is often widely dispersed by IoT networks that enable WSNs. Data duplication could occur if numerous sensor devices are used to monitor a location. Finding a solution to this issue by using clustering. Clustering lessens network traffic while retaining path dependability compared to the multipath technique. To relieve duplicate data in EECRPSID, we applied the clustering technique. The multipath strategy might make the provided protocol more dependable. Using the EECRPSID algorithm, will reduce the overall energy consumption, minimize the End-to-end delay to 0.14s, achieve a 99.8% Packet Delivery Ratio, and the network's lifespan will be increased. The NS2 simulator is used to run the whole set of simulations. The EECRPSID method has been implemented in NS2, and simulated results indicate that comparing the other three technologies improves the performance measures.
Analysis and Evolution of SHA-1 Algorithm - Analytical TechniqueIJCNCJournal
A 160-bit (20-byte) hash value, sometimes called a message digest, is generated using the SHA-1 (Secure Hash Algorithm 1) hash function in cryptography. This value is commonly represented as 40 hexadecimal digits. It is a Federal Information Processing Standard in the United States and was developed by the National Security Agency. Although it has been cryptographically cracked, the technique is still in widespread usage. In this work, we conduct a detailed and practical analysis of the SHA-1 algorithm's theoretical elements and show how they have been implemented through the use of several different hash configurations.
Optimizing CNN-BiGRU Performance: Mish Activation and Comparative AnalysisIJCNCJournal
Deep learning is currently extensively employed across a range of research domains. The continuous advancements in deep learning techniques contribute to solving intricate challenges. Activation functions (AF) are fundamental components within neural networks, enabling them to capture complex patterns and relationships in the data. By introducing non-linearities, AF empowers neural networks to model and adapt to the diverse and nuanced nature of real-world data, enhancing their ability to make accurate predictions across various tasks. In the context of intrusion detection, the Mish, a recent AF, was implemented in the CNN-BiGRU model, using three datasets: ASNM-TUN, ASNM-CDX, and HOGZILLA. The comparison with Rectified Linear Unit (ReLU), a widely used AF, revealed that Mish outperforms ReLU, showcasing superior performance across the evaluated datasets. This study illuminates the effectiveness of AF in elevating the performance of intrusion detection systems.
An Hybrid Framework OTFS-OFDM Based on Mobile Speed EstimationIJCNCJournal
The Future wireless communication systems face the challenging task of simultaneously providing high-quality service (QoS) and broadband data transmission, while also minimizing power consumption, latency, and system complexity. Although Orthogonal Frequency Division Multiplexing (OFDM) has been widely adopted in 4G and 5G systems, it struggles to cope with a significant delay and Doppler spread in high mobility scenarios. To address these challenges, a novel waveform named Orthogonal Time Frequency Space (OTFS). Designers aim to outperform OFDM by closely aligning signals with the channel behaviour. In this paper, we propose a switching strategy that empowers operators to select the most appropriate waveform based on an estimated speed of the mobile user. This strategy enables the base station to dynamically choose the waveform that best suits the mobile user’s speed. Additionally, we suggest retaining an Integrated Sensing and Communication (ISAC) radar approach for accurate Doppler estimation. This provides precise information to facilitate the waveform selection procedure. By leveraging the switching strategy and harnessing the Doppler estimation capabilities of an ISAC radar.Our proposed approach aims to enhance the performance of wireless communication systems in high mobility cases. Considering the complexity of waveform processing, we introduce an optimized hybrid system that combines OTFS and OFDM, resulting in reduced complexity while still retaining performance benefits.This hybrid system presents a promising solution for improving the performance of wireless communication systems in higher mobility.The simulation results validate the effectiveness of our approach, demonstrating its potential advantages for future wireless communication systems. The effectiveness of the proposed approach is validated by simulation results as it will be illustrated.
Enhanced Traffic Congestion Management with Fog Computing - A Simulation-Base...IJCNCJournal
Accurate latency computation is essential for the Internet of Things (IoT) since the connected devices generate a vast amount of data that is processed on cloud infrastructure. However, the cloud is not an optimal solution. To overcome this issue, fog computing is used to enable processing at the edge while still allowing communication with the cloud. Many applications rely on fog computing, including traffic management. In this paper, an Intelligent Traffic Congestion Mitigation System (ITCMS) is proposed to address traffic congestion in heavily populated smart cities. The proposed system is implemented using fog computing and tested in a crowdedCairo city. The results obtained indicate that the execution time of the simulation is 4,538 seconds, and the delay in the application loop is 49.67 seconds. The paper addresses various issues, including CPU usage, heap memory usage, throughput, and the total average delay, which are essential for evaluating the performance of the ITCMS. Our system model is also compared with other models to assess its performance. A comparison is made using two parameters, namely throughput and the total average delay, between the ITCMS, IOV (Internet of Vehicle), and STL (Seasonal-Trend Decomposition Procedure based on LOESS). Consequently, the results confirm that the proposed system outperforms the others in terms of higher accuracy, lower latency, and improved traffic efficiency.
Rendezvous Sequence Generation Algorithm for Cognitive Radio Networks in Post...IJCNCJournal
Recent natural disasters have inflicted tremendous damage on humanity, with their scale progressively increasing and leading to numerous casualties. Events such as earthquakes can trigger secondary disasters, such as tsunamis, further complicating the situation by destroying communication infrastructures. This destruction impedes the dissemination of information about secondary disasters and complicates post-disaster rescue efforts. Consequently, there is an urgent demand for technologies capable of substituting for these destroyed communication infrastructures. This paper proposes a technique for generating rendezvous sequences to swiftly reconnect communication infrastructures in post-disaster scenarios. We compare the time required for rendezvous using the proposed technique against existing methods and analyze the average time taken to establish links with the rendezvous technique, discussing its significance. This research presents a novel approach enabling rapid recovery of destroyed communication infrastructures in disaster environments through Cognitive Radio Network (CRN) technology, showcasing the potential to significantly improve disaster response and recovery efforts. The proposed method reduces the time for the rendezvous compared to existing methods, suggesting that it can enhance the efficiency of rescue operations in post-disaster scenarios and contribute to life-saving efforts.
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.
May 2024, Volume 16, Number 3 - The International Journal of Computer Network...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.
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.
[OReilly Superstream] Occupy the Space: A grassroots guide to engineering (an...Jason Yip
The typical problem in product engineering is not bad strategy, so much as “no strategy”. This leads to confusion, lack of motivation, and incoherent action. The next time you look for a strategy and find an empty space, instead of waiting for it to be filled, I will show you how to fill it in yourself. If you’re wrong, it forces a correction. If you’re right, it helps create focus. I’ll share how I’ve approached this in the past, both what works and lessons for what didn’t work so well.
zkStudyClub - LatticeFold: A Lattice-based Folding Scheme and its Application...Alex Pruden
Folding is a recent technique for building efficient recursive SNARKs. Several elegant folding protocols have been proposed, such as Nova, Supernova, Hypernova, Protostar, and others. However, all of them rely on an additively homomorphic commitment scheme based on discrete log, and are therefore not post-quantum secure. In this work we present LatticeFold, the first lattice-based folding protocol based on the Module SIS problem. This folding protocol naturally leads to an efficient recursive lattice-based SNARK and an efficient PCD scheme. LatticeFold supports folding low-degree relations, such as R1CS, as well as high-degree relations, such as CCS. The key challenge is to construct a secure folding protocol that works with the Ajtai commitment scheme. The difficulty, is ensuring that extracted witnesses are low norm through many rounds of folding. We present a novel technique using the sumcheck protocol to ensure that extracted witnesses are always low norm no matter how many rounds of folding are used. Our evaluation of the final proof system suggests that it is as performant as Hypernova, while providing post-quantum security.
Paper Link: https://eprint.iacr.org/2024/257
Introduction of Cybersecurity with OSS at Code Europe 2024Hiroshi SHIBATA
I develop the Ruby programming language, RubyGems, and Bundler, which are package managers for Ruby. Today, I will introduce how to enhance the security of your application using open-source software (OSS) examples from Ruby and RubyGems.
The first topic is CVE (Common Vulnerabilities and Exposures). I have published CVEs many times. But what exactly is a CVE? I'll provide a basic understanding of CVEs and explain how to detect and handle vulnerabilities in OSS.
Next, let's discuss package managers. Package managers play a critical role in the OSS ecosystem. I'll explain how to manage library dependencies in your application.
I'll share insights into how the Ruby and RubyGems core team works to keep our ecosystem safe. By the end of this talk, you'll have a better understanding of how to safeguard your code.
The Department of Veteran Affairs (VA) invited Taylor Paschal, Knowledge & Information Management Consultant at Enterprise Knowledge, to speak at a Knowledge Management Lunch and Learn hosted on June 12, 2024. All Office of Administration staff were invited to attend and received professional development credit for participating in the voluntary event.
The objectives of the Lunch and Learn presentation were to:
- Review what KM ‘is’ and ‘isn’t’
- Understand the value of KM and the benefits of engaging
- Define and reflect on your “what’s in it for me?”
- Share actionable ways you can participate in Knowledge - - Capture & Transfer
Session 1 - Intro to Robotic Process Automation.pdfUiPathCommunity
👉 Check out our full 'Africa Series - Automation Student Developers (EN)' page to register for the full program:
https://bit.ly/Automation_Student_Kickstart
In this session, we shall introduce you to the world of automation, the UiPath Platform, and guide you on how to install and setup UiPath Studio on your Windows PC.
📕 Detailed agenda:
What is RPA? Benefits of RPA?
RPA Applications
The UiPath End-to-End Automation Platform
UiPath Studio CE Installation and Setup
💻 Extra training through UiPath Academy:
Introduction to Automation
UiPath Business Automation Platform
Explore automation development with UiPath Studio
👉 Register here for our upcoming Session 2 on June 20: Introduction to UiPath Studio Fundamentals: https://community.uipath.com/events/details/uipath-lagos-presents-session-2-introduction-to-uipath-studio-fundamentals/
"Scaling RAG Applications to serve millions of users", Kevin GoedeckeFwdays
How we managed to grow and scale a RAG application from zero to thousands of users in 7 months. Lessons from technical challenges around managing high load for LLMs, RAGs and Vector databases.
Northern Engraving | Modern Metal Trim, Nameplates and Appliance PanelsNorthern Engraving
What began over 115 years ago as a supplier of precision gauges to the automotive industry has evolved into being an industry leader in the manufacture of product branding, automotive cockpit trim and decorative appliance trim. Value-added services include in-house Design, Engineering, Program Management, Test Lab and Tool Shops.
How information systems are built or acquired puts information, which is what they should be about, in a secondary place. Our language adapted accordingly, and we no longer talk about information systems but applications. Applications evolved in a way to break data into diverse fragments, tightly coupled with applications and expensive to integrate. The result is technical debt, which is re-paid by taking even bigger "loans", resulting in an ever-increasing technical debt. Software engineering and procurement practices work in sync with market forces to maintain this trend. This talk demonstrates how natural this situation is. The question is: can something be done to reverse the trend?
Have you ever been confused by the myriad of choices offered by AWS for hosting a website or an API?
Lambda, Elastic Beanstalk, Lightsail, Amplify, S3 (and more!) can each host websites + APIs. But which one should we choose?
Which one is cheapest? Which one is fastest? Which one will scale to meet our needs?
Join me in this session as we dive into each AWS hosting service to determine which one is best for your scenario and explain why!
In the realm of cybersecurity, offensive security practices act as a critical shield. By simulating real-world attacks in a controlled environment, these techniques expose vulnerabilities before malicious actors can exploit them. This proactive approach allows manufacturers to identify and fix weaknesses, significantly enhancing system security.
This presentation delves into the development of a system designed to mimic Galileo's Open Service signal using software-defined radio (SDR) technology. We'll begin with a foundational overview of both Global Navigation Satellite Systems (GNSS) and the intricacies of digital signal processing.
The presentation culminates in a live demonstration. We'll showcase the manipulation of Galileo's Open Service pilot signal, simulating an attack on various software and hardware systems. This practical demonstration serves to highlight the potential consequences of unaddressed vulnerabilities, emphasizing the importance of offensive security practices in safeguarding critical infrastructure.
Skybuffer SAM4U tool for SAP license adoptionTatiana Kojar
Manage and optimize your license adoption and consumption with SAM4U, an SAP free customer software asset management tool.
SAM4U, an SAP complimentary software asset management tool for customers, delivers a detailed and well-structured overview of license inventory and usage with a user-friendly interface. We offer a hosted, cost-effective, and performance-optimized SAM4U setup in the Skybuffer Cloud environment. You retain ownership of the system and data, while we manage the ABAP 7.58 infrastructure, ensuring fixed Total Cost of Ownership (TCO) and exceptional services through the SAP Fiori interface.
Connector Corner: Seamlessly power UiPath Apps, GenAI with prebuilt connectorsDianaGray10
Join us to learn how UiPath Apps can directly and easily interact with prebuilt connectors via Integration Service--including Salesforce, ServiceNow, Open GenAI, and more.
The best part is you can achieve this without building a custom workflow! Say goodbye to the hassle of using separate automations to call APIs. By seamlessly integrating within App Studio, you can now easily streamline your workflow, while gaining direct access to our Connector Catalog of popular applications.
We’ll discuss and demo the benefits of UiPath Apps and connectors including:
Creating a compelling user experience for any software, without the limitations of APIs.
Accelerating the app creation process, saving time and effort
Enjoying high-performance CRUD (create, read, update, delete) operations, for
seamless data management.
Speakers:
Russell Alfeche, Technology Leader, RPA at qBotic and UiPath MVP
Charlie Greenberg, host
Dandelion Hashtable: beyond billion requests per second on a commodity serverAntonios Katsarakis
This slide deck presents DLHT, a concurrent in-memory hashtable. Despite efforts to optimize hashtables, that go as far as sacrificing core functionality, state-of-the-art designs still incur multiple memory accesses per request and block request processing in three cases. First, most hashtables block while waiting for data to be retrieved from memory. Second, open-addressing designs, which represent the current state-of-the-art, either cannot free index slots on deletes or must block all requests to do so. Third, index resizes block every request until all objects are copied to the new index. Defying folklore wisdom, DLHT forgoes open-addressing and adopts a fully-featured and memory-aware closed-addressing design based on bounded cache-line-chaining. This design offers lock-free index operations and deletes that free slots instantly, (2) completes most requests with a single memory access, (3) utilizes software prefetching to hide memory latencies, and (4) employs a novel non-blocking and parallel resizing. In a commodity server and a memory-resident workload, DLHT surpasses 1.6B requests per second and provides 3.5x (12x) the throughput of the state-of-the-art closed-addressing (open-addressing) resizable hashtable on Gets (Deletes).
"Choosing proper type of scaling", Olena SyrotaFwdays
Imagine an IoT processing system that is already quite mature and production-ready and for which client coverage is growing and scaling and performance aspects are life and death questions. The system has Redis, MongoDB, and stream processing based on ksqldb. In this talk, firstly, we will analyze scaling approaches and then select the proper ones for our system.
5th LF Energy Power Grid Model Meet-up SlidesDanBrown980551
5th Power Grid Model Meet-up
It is with great pleasure that we extend to you an invitation to the 5th Power Grid Model Meet-up, scheduled for 6th June 2024. This event will adopt a hybrid format, allowing participants to join us either through an online Mircosoft Teams session or in person at TU/e located at Den Dolech 2, Eindhoven, Netherlands. The meet-up will be hosted by Eindhoven University of Technology (TU/e), a research university specializing in engineering science & technology.
Power Grid Model
The global energy transition is placing new and unprecedented demands on Distribution System Operators (DSOs). Alongside upgrades to grid capacity, processes such as digitization, capacity optimization, and congestion management are becoming vital for delivering reliable services.
Power Grid Model is an open source project from Linux Foundation Energy and provides a calculation engine that is increasingly essential for DSOs. It offers a standards-based foundation enabling real-time power systems analysis, simulations of electrical power grids, and sophisticated what-if analysis. In addition, it enables in-depth studies and analysis of the electrical power grid’s behavior and performance. This comprehensive model incorporates essential factors such as power generation capacity, electrical losses, voltage levels, power flows, and system stability.
Power Grid Model is currently being applied in a wide variety of use cases, including grid planning, expansion, reliability, and congestion studies. It can also help in analyzing the impact of renewable energy integration, assessing the effects of disturbances or faults, and developing strategies for grid control and optimization.
What to expect
For the upcoming meetup we are organizing, we have an exciting lineup of activities planned:
-Insightful presentations covering two practical applications of the Power Grid Model.
-An update on the latest advancements in Power Grid -Model technology during the first and second quarters of 2024.
-An interactive brainstorming session to discuss and propose new feature requests.
-An opportunity to connect with fellow Power Grid Model enthusiasts and users.
Real time audio translation module between iax and rsw
1. International Journal of Computer Networks & Communications (IJCNC) Vol.6, No.3, May 2014
DOI : 10.5121/ijcnc.2014.6310 121
REAL-TIME AUDIO TRANSLATION MODULE
BETWEEN IAX AND RSW
Hadeel Saleh Haj Aliwi and Putra Sumari
School of Computer Sciences, Universiti Sains Malaysia, Penang, Malaysia
ABSTRACT
At the last few years, multimedia communication has been developed and improved rapidly in order to
enable users to communicate between each other over the internet. Generally, multimedia communication
consists of audio and video communication. However, this research concentrates on audio conferencing
only. The audio translation between protocols is a very critical issue, because it solves the communication
problems between any two protocols. So, it enables people around the world to talk with each other even
they use different protocols. In this research, a real time audio translation module between two protocols
has been done. These two protocols are: InterAsterisk eXchange Protocol (IAX) and Real-Time Switching
Control Protocol (RSW), which they are widely used to provide two ways audio transfer feature. The
solution here is to provide interworking between the two protocols which they have different media
transports, audio codec’s, header formats and different transport protocols for the audio transmission. This
translation will help bridging the gap between the two protocols by providing interworking capability
between the two audio streams of IAX and RSW. Some related works have been done to provide translation
between IAX and RSW control signalling messages. But, this research paper concentrates on the
translation that depends on the media transfer. The proposed translation module was tested and evaluated
in different scenarios in order to examine its performance. The obtained results showed that the Real-Time
Audio Translation Module produces lower rates of packet delay and jitter than the acceptance values for
each of the mentioned performance metrics.
KEYWORDS
Multimedia, VoIP, Interworking, InterAsterisk eXchange Protocol (IAX), Real Time Switching Control
Criteria (RSW).
1. INTRODUCTION
Multimedia communication has been developed and improved to be the basic and essential
service in order to satisfy the needs of the Internet users [12, 18, 19, 20]. It has appeared to be
more and more applicable in distributed environments [10, 21]. There are various protocols that
control and signal the calls of the Internet telephone [1]; such protocols existed in Internet
Protocol (IP) telecommunication. For data and signaling, two major protocols are considered in
the field of the multimedia conferencing such as RSW and IAX protocols. Video functionality,
similar quality of service and competitive voice are provided by these two protocols. The two
protocols have been widely exploited and utilized into many different methods [15]. This work
proposes a translation module between audio streams of IAX protocol and RSW control protocol.
Generally, the procedure to a translator between the both protocols is starting from the first
network to perform transferring the data into a first protocol. And then the second network to
perform transferring the data into a second protocol. Finally, the translation server (conference
gateway) to maintain translation information for a protocol translation, this translation occurs
between the first and the second protocols [15].
2. International Journal of Computer Networks & Communications (IJCNC) Vol.6, No.3, May 2014
122
IAX and RSW audio transfer protocols have many differences in handling and exchanging voice
packets between each other. In addition, different transport protocols and techniques are used by
each of these protocols throughout audio exchange. The packet format in IAX protocol differs
from the corresponding packet in RSW control protocol. When the data is attempted to be
exchanged among each other by the two protocols, the data could not be understood by each of
the two protocols. To overcome the translation problem between the audio streams of the two
protocols (IAX and RSW), we proposed an audio translation module, which will be able to act as
a translator between IAX and RSW audio streams. This translation module will help bridging the
gap between IAX and RSW by providing interworking capability between the two audio streams.
There are two sides accompanied with the translation between the two protocols: the control
signaling and audio stream translation. The control signaling translation between two protocols,
more specifically between RSW control protocol and IAX protocol has already done [6,7]. And
the audio stream translation, which is the proposed work in this paper. The performed audio
stream translation is true while this research concentrates on the two protocols (RSW control
protocol and IAX protocol).
2. BRIEF DESCRIPTION OF IAX AND RSW PROTOCOLS
This section will briefly describe the both protocols; Real-time Switching (RSW) Control
Protocol and InterAsterisk eXchange (IAX) Protocol.
2.1. RSW Control Protocol
Real-time Switching (RSW) control criteria is a control protocol used to handle a multipoint-to-
multipoint multimedia conferencing sessions. RSW control protocol was developed in 1993 as a
control mechanism for conferencing by the Network Research Group in school of computer
sciences, University Sciences Malaysia (USM) [9]. RSW protocol doesn’t have its own header to
carry the data so, Real-Time Protocol (RTP) protocol [3] is used to carry audio and video data
through multimedia conferencing. User Datagram Protocol (UDP) is a transport protocol [2] that
is also used by RSW to transfer audio and video data. The RSW control criteria is involved in
decreasing bandwidth when many clients using the MCS system. RSW makes a list of priority for
the participants to avoid confusion when many participants are trying to speak up during
conference [6,13]. There are several advantages for the RSW control criteria [9] such as Equal
Privileges, First Come First Serve, First come first serve with time-out, Organizer Main Site and
Restricted Active site. The RSW audio packet format consists of four parts as shown in figure 1,
which are IP header [2], UDP header, RTP header, and and RSW payload (varies based on the
codec used) which is the body of the audio message.
IP (20 bytes) UDP (8 bytes) RTP (12 bytes) RSW Payload
Figure 1. RSW Audio Packet Format.
2.2. InterAsterisk eXchange Protocol
In (2004) Mark Spencer [5] has created the Inter-Asterisk eXchange (IAX) protocol for asterisk
that performs VoIP signalling. Streaming media is managed, controlled and transmitted through
the Internet Protocol (IP) networks based on this protocol. Any type of streaming media could be
3. International Journal of Computer Networks & Communications (IJCNC) Vol.6, No.3, May 2014
123
used by this protocol. However, IP voice calls are basically being controlled by IAX protocol
[14]. Furthermore, this protocol can be called as a peer to peer (P2P) protocol that performs two
types of connections which are Voice over IP (VoIP) connections through the servers and Client-
Server communication. IAX is currently changed to IAX2 which is the second version of the IAX
protocol. The IAX2 has deprecated the original IAX protocol [5]. Call signalling and multimedia
transport functions are supported by the IAX protocol. In the same session and by using IAX,
Voice streams (multimedia and signalling) are conveyed. Furthermore, IAX supports the trunk
connections concept for numerous calls. The bandwidth usage is reduced when this concept is
being used because all the protocol overhead is shared for all the calls between two IAX nodes.
Over a single link, IAX provides multiplexing channels [11]. IAX is a simple protocol in such a
way Network Address Translation (NAT) traversal complications are avoided by it [8]. Unlike
RSW, IAX has its own frames (full and mini frames) to carry control, audio, video messages and
signals. IAX full frames are responsible to carry the signals and the control messages, while Mini
frame are the only type of frames which hold the data during the media transmission session
between two endpoints A and B. Each audio/video flow is of IAX Mini Frames (M frames)
contains 4 bytes header (source call number (2 bytes) and timestamp (2 bytes)). UDP transport
protocol is used by IAX to transfer audio and video data [4]. The IAX audio packet format
consists of four parts, which are the IP header, the UDP header, the IAX header [5] and the IAX
payload (varies based on the codec used). Audio packet format of the IAX protocol is shown in
figure 2.
IP (20 bytes) UDP (8 bytes) IAX header (4 bytes) IAX payload
Figure 2. IAX Audio Packet Format.
3. CODEC’S USED IN IAX-RSW TRANSLATION MODULE
In order to transmit the packets from the sender to the receiver, an audio codec should be used. In
this paper, GSM codec is proposed in order to convert the analog audio to digital form [17]. Then,
the digital audio will be compressed to decrease the consumption of network bandwidth needed to
pass on the speech to the receiver. GSM audio codec has the bandwidth of (13.2 kb/s). However,
while sending the codec data across an IP network, it is going to increase the used bandwidth.
While dealing with voice, it is better not to introduce too much latency. For example, sending
voice frames every 20 ms, which means every 50 frames have to be sent in one second. Dividing
13.2 kb/s by 50 will give 33 bytes. As a result, 33 bytes of voice data will be sent per one frame.
In addition, IP header (20 bytes a packet), UDP header (8 bytes a packet) adding the RTP
protocol header (12 bytes a packet). This codec will be used not only for the IAX protocol, but
also for the RSW control protocol. In this paper, the two audio streams protocols will use the
same codec to transmit and receive the audio packets between the clients.
4. IAX-RSW TRANSLATION METHOD
This section will explain the two audio streams translation methods, which are RSW to IAX and
IAX to RSW. The proposed translation module consists of three components: IAX network, RSW
network, and the translation server or it can be called conference gateway. The proposed
conference gateway includes two buffers; IAX to RSW buffer and RSW to IAX buffer. The main
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task of these buffers is converting the packet format of the source protocol to be identical to the
format of the destination protocol to be clearly received.
• IAX to RSW buffer:
1. Collecting and storing the packets received from IAX client.
2. Extracting the mini header from IAX packets.
3. Inserting the RTP header (on behalf of the mini header) to IAX packets in order to be
identical to RSW packet format.
4. Preparing the converted packets to be sent to the RSW client.
• RSW to IAX buffer:
1. Collecting and storing the packets received from RSW client.
2. Extracting the RTP header from RSW packets.
3. Inserting the mini header (on behalf of RTP header) to RSW packets in order to be
identical to IAX packet format.
4. Preparing the converted packets to be sent to the IAX client.
5. SIMULATION ENVIRONMENT
Generally, there are no objections to have the flexibility to choose any simulation program to
simulate the proposed translation module as long as it meets the requirements to simulate such a
service. Furthermore, there are no restrictions to use particular hardware as a base to simulate the
translation module. GloMoSim (Global Mobile system Simulator) was used to simulate the
translation module [23]. GloMoSim is preferred to be chosen because it is extensible and
composable [22]. The operating system that was used as an environment to run GloMoSim is
Windows Vista. The hardware platform that was used to simulate the proposed translation module
is Intel Core Duo 2.00 GHz.
The simulation model consists of three main elements, which are: IAX network, RSW network,
and the conference gateway gateways with the internet connection. In the proposed simulation
model, voice packets are collected in the gateway’s buffer. The translation processes are done in
the gateway. This simulation model has many parameters in order to understand the simulation
processing, such as number of clients, buffer size, audio codec...etc.
6. PERFORMANCE METRICS TESTS
The real-time audio streams of IAX-RSW translation module will be evaluated by measuring and
testing by two performance metrics, which are: Packet Delay and Jitter. In this scenarios, the
packet delay and jitter tests of IAX-RSW translation module (one-to-one) were performed by
taking one RSW client and one IAX client. Both ways (RSW-to-IAX and IAX-to-RSW) were
tested using variable packet sequence numbers ranging from 10 to 100 packets.
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6.1. Packet Delay Test
Packet Delay is defined as “The measure of time that it takes the talker’s voice to reach the
listener’s ear” [24]. In the real-time applications, voice delay is considered to be a critical issue,
such as audio conferencing systems [25].
The end-to-end acceptable delay is 150 milli secondsPacket delay examines the quality of VoIP
service; as packet delay increased the QoS of VoIP calls decreased.
Packet delay is ranged between 1 and 15 ms in RSW-to-IAX transmission, while packet delay is
ranged between 1 and 12 ms in IAX-to-RSW transmission.
As IAX protocol has trunking property, this makes the quality of services for this protocol are
better than the other session protocols. Figure 3 shows that the IAX-RSW translation module
result values for packet delay are less than the accepted packet delay value which is 150 ms.
6.2. Jitter Test
The transmitted packets from the source to the destination may be reached with different delays
that rely on the queue in the crowded networks, several paths to avoid congestion, etc. This case
cause dissimilar time variations.
The different time variations of received packets on the destination are known as Jitter. Jitter can
bother and influence the quality of audio. Jitter should be less than 30 milliseconds [26].
Jitter examines the quality of VoIP service; as jitter increased the QoS of VoIP calls decreased.
According to figure 4, jitter is ranged between 2 and 4.3 ms in RSW-to-IAX transmission, while
it is ranged between 2 and 3.9 ms in IAX-to-RSW transmission.
We can notice that the jitter results of both ways are almost convergent.
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Figure 4. One-to-One IAX-RSW Packet Delay Results.
Figure 5. One-to-One IAX-RSW Jitter Results.
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7. SUMMARY
For guaranteeing a seamless end to end connectivity for IAX-RSW translation, we have proposed
solution to these interworking problems by doing an audio translation module. Using the
proposed Translation Module will enable VoIP protocols uses different sizes and formats of data
chunks to exchange audio streams. The translation module can be a base for any two different
protocols or more. The proposed method came as a translation audio streams module that can be
used when provide IAX-RSW full interworking. It will be a valuable research if the translation
module that depends on media transfer part (which is this performed work) combined with
translation module that depends on the signalling part, so we can get completed results. The
proposed translation module was tested and evaluated in different scenarios in order to examine
its performance. The obtained results showed that the Real-Time Translation Module produces
lower rates of packet delay and jitter than the acceptance values for each of the mentioned
performance metrics.
REFERENCES
[1] A. Uyar, (2005) “Scalable Oriented Architecture for Audio/Video conferencing,” Ph.D. Thesis,
Syracuse University, USA.
[2] A.S. Tanenbaum, (2003) “Computer Networks,” 4th edition, Pearson Education, Inc.
[3] C. Perkins, (2003) “RTP: Audio and Video for the Internet,” Addison Wesley, USA.
[4] D. DiNicolo, (2007) “Transporting VoIP Traffic with UDP and RTP”.
[5] M. Spencer & F.W. Miller, (2004) “IAX Protocol Description”.
[6] M.S. Kolhar, A.F. Bayan, T.C. Wan, O. Abouabdalla & R. Sureswaran, (2008) “Control and Media
Session: IAX with RSW Control Criteria,” Proceedings of International Conference on Network
Applications, Protocols and Services, Executive Development Centre, Universiti Utara Malaysia,
pp.130-135.
[7] M.S. Kolhar, A.F. Bayan, T.C. Wan, O. Abouabdalla & R. Sureswaran, (2008) “Multimedia
Communication: RSW Control Protocol and IAX,” The 5th International Symposium on High
Capacity Optical Networks and Enabling Technologies, Penang, Malaysia, pp.75-79.
[8] M.S. Kolhar, M.M. Abu-Alhaj, O. Abouabdalla, T.C. Wan & A.M. Manasrah, (2009) “Comparative
Evaluation and Analysis of IAX and RSW,” International Journal of Computer Science and
Information Security, pp.250-252.
[9] O. Abouabdalla & R. Sureswaran, (2006) “Enable Communication between the RSW Control Criteria
and SIP Using R2SP,” The 2nd International Conference on Distributed Frameworks for Multimedia
Applications, pp.1-7.
[10] O. Abouabdalla & R. Sureswaran, (2000) “Server Algorithm to Manage Distributed Network Entities
for Multimedia Conferencing System,” In Proceedings IWS Asia Pacific Advanced Network and its
Applications, pp.141-146.
[11] P. Montoro & E. Casilari, (2009) “A Comparative Study of VoIP Standards with Asterisk,”
Proceedings of the Fourth International Conference on Digital Telecommunications, pp.1-6.
[12] S. Gale, (1990) “Human Aspect of Interactive Multimedia Communication Interaction with
Computers”.
8. International Journal of Computer Networks & Communications (IJCNC) Vol.6, No.3, May 2014
128
[13] R. Sureswaran, R.K. Subramanian, H. Guyennet & M. Trehel, (1997) “Using RSW Control
Criteria to Create a Distributed Environment for Multimedia Conferencing,” Proceedings on Research
and Development in Computer Science and its Applications, Penang, Malaysia, pp.27-29.
[14] T. Abbasi, S. Prasad, N. Seddigh & I. Lambadaris, (2005) “A Comparative Study of the SIP and
IAX,” Canadian Conference on Electrical and Computer Engineering, pp.179- 183.
[15] T. Oishi & H. Inouchi, (2007) “Method and System for Persistent Translation between Protocols”.
[16] T. Kelliher, (2008) “TCP and UDP socket programming”.
[17] V. Toncar, (2009) “VoIP Basics: Overview of Audio Codec’s”.
[18] Y. Wan, X. Zhang & W. Li, (2000) “Audio Multimedia Conferencing System Based on the
Technology of Speech Recognition,” The 2000 IEEE Asia-Pacific Conference on Circuits and
Systems, pp.771-774.
[19] A. Uyar, S. Pallickara & G. Fox, (2003) “Towards an Architecture for Audio/Video Conferencing in
Distributed Brokering Systems,” Proceedings of the International Conference on Communications in
Computing, Las Vegas, USA, pp.17-23.
[20] S.A.K. Al-Omari, P. Sumari, S. Al-Taweel & A.M. Manasrah, (2010) “CUSTP: Custom Protocol for
Audio and Video Conferencing System over P2P Networks,” International Journal of Digital Content
Technology and its Applications, pp.61-74.
[21] B. White, J. Lepreau, L. Stoller, R. Ricci, S. Guruprasad, M. Newbold, M. Hibler, C. Barb, & A.
Joglekar, (2002) “An Integrated Experimental Environment for Distributed Systems and Networks,”
in Proceeding of the Fifth Symposium on Operating Systems Design and Implementation, Salt Lake
City, Utah 84112-9205 USA, pp.255-270.
[22] X. Zeng, R. Bagrodia & M. Gerla: GloMoSim, (1998) “A Library for Parallel Simulation of Large-
scale Wireless Networks,” Proceeding of the IEEE PADS 98 Parallel and distributed simulation,
Banff, Alta, pp.154-161.
[23] J. Hsu, T. Yan & R. Bagrodia, (1999) “A Simulation Study of Packetized Voice Over Multi-hop
Wireless Networks using GloMoSim,” Proceedings of the International Conference on Parallel and
Distributed Processing Techniques and Applications, Las Vegas, Nevada, USA.
[24] G. Rittenhouse & H. Zheng, (2005) “Providing VOIP Service in UMTS-HSDPA with Frame
Aggregation,” Proceeding IEEE International Conference on Acoustics, Speech, and Signal
Processing, pp. 1157-1160.
[25] D. Su & J. Srivastava, (1999) “Investigating factors influencing QoS of internet phone, Proceeding
International Conference on Multimedia Computing and Systems, pp. 308-313.
[26] A.R. Reibman, V.A. Vaishampayan & Y. Sermadevi, (2004) “Quality Monitoring of Video Over a
Packet Network,” IEEE Transactions, pp. 327- 334.
9. International Journal of Computer Networks & Communications (IJCNC) Vol.6, No.3, May 2014
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Authors
Hadeel Saleh Haj Aliwi has obtained her Bachelor degree in Computer Engineering from
Ittihad Private University, Syria in 2007-2008 and Master degree in Computer Science from
Universiti Sains Malaysia, Penang, Malaysia in 2011. Currently, she is a PhD candidate at the
School of Computer Science, Universiti Sains Malaysia. Her main research area interests are in
includes Multimedia Networking, Voice over Internet protocols (VoIP), and Real-time
Interworking between Heterogeneous signalling protocols, and Instant Messaging protocols.
Putra Sumari obtained his MSc and PhD in 1997 and 2000 from Liverpool University,
England. Currently, he is Associate Professor and a lecturer at the School of Computer Science,
USM. He is the head of the Multimedia Computing Research Group, CS, USM. Member of
ACM and IEEE, Program Committee and reviewer of several International Conference on
Information and Communication Technology (ICT), Committee of Malaysian ISO Standard
Working Group on Software Engineering Practice, Chairman of Industrial Training Program,
School of Computer Science, USM, Advisor of Master in Multimedia Education Program, UPSI, Perak.