The aim of this paper is to present the ‘Power’ of SDN Technology and MEC Technic in improving the delivering of IPTV Service. Those days, the IPTV end –users are tremendous increased all over the world , but in the same time also the complains for receiving these prepaid real time multimedial services like; high latency, high bandwidth, low performance and low QoE/QoS. On the other end, IPTV Distributors need a new system, technics, network solutions to distribute content continuesly and simultaneously to all active end-users with high-quality, lowlatency and high Performance, thus monitoring and re-configuring this ‘Big Data’ require high Bandwidth by causing difficult problems by offering it affecting in the same time the price and QoE/QoSperformance of delivered service.
For this reason, we have achieved to optimize the IPTV service by applying SDN solution in a MEC Architecture (Multiple-Access Edge Computing). In this way , through MEC Technology and SDN, it is possible to receive an IPTV service with Low Latency, High Performance and Low Bandwidth by solving successfully all the problems faced by the actual IPTV Operators. These improvements of delivering IPTV service through MEC will be demonstrated by using the OMNet +++ simulator in an LTE-A mobile network. The results show clearly that by applying the MEC technique in the LTE-A network for receiving IPTV Service through SDN Network, the service was delivered with latency decreased by >90% (compared to the cases when the MEC technique is not applied), with PacketLoss of almost 0 and with high performance QoE. In addition these strong Contributions, the ‘Big’ innovation achieved in this work through simulations is that the quality of delivered IPTV Service did not change according to the increasing of the end-users.This latency of delivering the video streaming services did not change. This means that the IPTV Service providers will increase their benefits by ensuring in the same time also the delivering of service with high quality and performance toward innumerous end users. Consequently, MEC Technology and SDN solution will be the two right and "smart" network choices that will boost the development of the 5th Mobile generation and will significantly improve the benefit of Video Streaming services offered by current providers worldwide (Netflix, HULU, Amazon Prime, YouTube, etc).
The leading method of correspondence is clearly through voice trade. There are essentially two different ways through which voice can be effortlessly communicated on an organization: PSTN (Public Switched Telephone Network) and VoIP (Voice over Internet Protocol).
Mainly represented by SIP, VoIP protocols and implementations contain several vulnerabilities, particularly related to their complexities and in the face of interoperability of telephony equipment’s.
It was by identifying a lack of literature with focus in security and potential vulnerabilities of the SIP Protocol that we propose in this document. We attempt to provide a theoretical analysis from security aspects used by one of the signaling call protocols, Session Initiation Protocol (SIP).
It is intended to lucidly illustrate and identify threats, vulnerabilities, security mechanisms, developed methods and protocols and, finally over time improvements.
Current trends and innovations in voice over IPALTANAI BISHT
Learn how to implement an open-source webrtc Click to dial or VOIP setup for their enterprises and also the new innovative add-on tech available for a basic VOIP system such as auto-attendants.
VoIP vs Telecom Providers
SIP Servers types
Open-source tool and technologies in VOIP
Opensip
Kamailio
Freeswitch
Media Handling
Webrtc
Machine learning in VoIP
Call Classifier
Fraud Detector
NLP and Auto attendants
VoIP to telecom bridging
MAF ICIMS™ Monitoring, Analytics & Reporting for Microsoft Teams and UC - glo...MAF InfoCom
MAF ICIMS™ is a reporting and analytics solution for Unified Communication and Collaboration (UC&C) platforms and other data sources such as Session Border Controllers (SBC’s), Gateways, Trading Platforms, Turrents & Dealer Boards. It allows you to gain valuable business and technical insights through its reports, daily dashboards and historical trend monitors. Its flexible, user defined nature means you tell the software what you want to see instead of the software dictating to you what you will see.
Introduction to Wireless cellular technologie and NGN,IMS ganeshmaali
This document provides an overview of wireless cellular technologies and introduces Next Generation Networks (NGN). It discusses 2G technologies like GSM and CDMA and how they focused on circuit switched voice, SMS, and low-speed data. It then covers 2.5G and 3G technologies like GPRS, EDGE, UMTS, CDMA2000, which enabled higher-speed packet-switched data. The document also discusses 4G technologies like LTE and LTE-Advanced, along with key aspects of their network architectures. Finally, it provides a brief introduction to NGN and the IP Multimedia Subsystem.
This document outlines the basic design of a Tier 3 ISP network. It defines key ISP terminology like tiers and points of presence. The project goal is described as providing internet access to 3 departments on campus through a wired or wireless network. The core architecture includes a backbone with a core router connecting to distribution routers that provide access to the departments. The document also covers addressing, routing protocols and how the network would be set up and tested.
This document appears to be a presentation on next generation networks and related topics. It includes sections on topics like passive optical networks versus Ethernet, MPLS VPNs, quality of service, IPv6 transition technologies, and network optimization approaches. The document contains diagrams, tables, and questions/answers related to these technical subject areas.
Netflix over Qos Enabled LTE Research Paper FinalAjit Kahaduwe
1) The document discusses how using Quality of Service (QoS) mechanisms can enhance the experience of streaming video applications like Netflix over cellular networks.
2) Tests were conducted on an LTE network to analyze how giving Netflix higher priority through QoS impacted video quality and buffering during congestion compared to default "best effort" treatment.
3) The results showed that higher QoS priority for Netflix improved video quality and reduced buffering events during congestion compared to best effort, demonstrating how QoS can provide a better user experience for video streaming.
The document discusses ITU initiatives on next generation networks and the way forward. It provides an overview of ITU standardization work on NGN, including developing standards and recommendations. It also discusses ITU's developmental work on NGN through various programs and initiatives aimed at promoting NGN adoption. This includes efforts related to regulatory reforms, capacity building, and case studies. The document outlines ITU's vision of broadband as critical infrastructure and highlights its economic and social benefits.
The leading method of correspondence is clearly through voice trade. There are essentially two different ways through which voice can be effortlessly communicated on an organization: PSTN (Public Switched Telephone Network) and VoIP (Voice over Internet Protocol).
Mainly represented by SIP, VoIP protocols and implementations contain several vulnerabilities, particularly related to their complexities and in the face of interoperability of telephony equipment’s.
It was by identifying a lack of literature with focus in security and potential vulnerabilities of the SIP Protocol that we propose in this document. We attempt to provide a theoretical analysis from security aspects used by one of the signaling call protocols, Session Initiation Protocol (SIP).
It is intended to lucidly illustrate and identify threats, vulnerabilities, security mechanisms, developed methods and protocols and, finally over time improvements.
Current trends and innovations in voice over IPALTANAI BISHT
Learn how to implement an open-source webrtc Click to dial or VOIP setup for their enterprises and also the new innovative add-on tech available for a basic VOIP system such as auto-attendants.
VoIP vs Telecom Providers
SIP Servers types
Open-source tool and technologies in VOIP
Opensip
Kamailio
Freeswitch
Media Handling
Webrtc
Machine learning in VoIP
Call Classifier
Fraud Detector
NLP and Auto attendants
VoIP to telecom bridging
MAF ICIMS™ Monitoring, Analytics & Reporting for Microsoft Teams and UC - glo...MAF InfoCom
MAF ICIMS™ is a reporting and analytics solution for Unified Communication and Collaboration (UC&C) platforms and other data sources such as Session Border Controllers (SBC’s), Gateways, Trading Platforms, Turrents & Dealer Boards. It allows you to gain valuable business and technical insights through its reports, daily dashboards and historical trend monitors. Its flexible, user defined nature means you tell the software what you want to see instead of the software dictating to you what you will see.
Introduction to Wireless cellular technologie and NGN,IMS ganeshmaali
This document provides an overview of wireless cellular technologies and introduces Next Generation Networks (NGN). It discusses 2G technologies like GSM and CDMA and how they focused on circuit switched voice, SMS, and low-speed data. It then covers 2.5G and 3G technologies like GPRS, EDGE, UMTS, CDMA2000, which enabled higher-speed packet-switched data. The document also discusses 4G technologies like LTE and LTE-Advanced, along with key aspects of their network architectures. Finally, it provides a brief introduction to NGN and the IP Multimedia Subsystem.
This document outlines the basic design of a Tier 3 ISP network. It defines key ISP terminology like tiers and points of presence. The project goal is described as providing internet access to 3 departments on campus through a wired or wireless network. The core architecture includes a backbone with a core router connecting to distribution routers that provide access to the departments. The document also covers addressing, routing protocols and how the network would be set up and tested.
This document appears to be a presentation on next generation networks and related topics. It includes sections on topics like passive optical networks versus Ethernet, MPLS VPNs, quality of service, IPv6 transition technologies, and network optimization approaches. The document contains diagrams, tables, and questions/answers related to these technical subject areas.
Netflix over Qos Enabled LTE Research Paper FinalAjit Kahaduwe
1) The document discusses how using Quality of Service (QoS) mechanisms can enhance the experience of streaming video applications like Netflix over cellular networks.
2) Tests were conducted on an LTE network to analyze how giving Netflix higher priority through QoS impacted video quality and buffering during congestion compared to default "best effort" treatment.
3) The results showed that higher QoS priority for Netflix improved video quality and reduced buffering events during congestion compared to best effort, demonstrating how QoS can provide a better user experience for video streaming.
The document discusses ITU initiatives on next generation networks and the way forward. It provides an overview of ITU standardization work on NGN, including developing standards and recommendations. It also discusses ITU's developmental work on NGN through various programs and initiatives aimed at promoting NGN adoption. This includes efforts related to regulatory reforms, capacity building, and case studies. The document outlines ITU's vision of broadband as critical infrastructure and highlights its economic and social benefits.
This document provides a summary of Prateek's professional experience in software development for telecom and networking. Over 9.5 years, he has worked on projects involving optical networking, load balancing servers, protocol development, and customer support. His responsibilities have included technical lead roles, individual development work, design, testing, and system integration. He has strong skills in C, C++, Linux, networking protocols, data structures, and development tools like version control systems. His work experience includes roles at NEC Technology, Brocade Communication, Juniper Networks, and Huawei Technology where he contributed to projects involving network security, load balancing, network address translation, and more.
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.
This document provides an overview of Denial of Service (DoS) attacks on Session Initiation Protocol (SIP) based Voice over Internet Protocol (VoIP) infrastructure. It first introduces VoIP and SIP, describing SIP components and messages. It then discusses security issues with SIP such as eavesdropping, message tampering, and spoofing. Several types of SIP DoS attacks are classified, including SIP message payload tampering, SIP message flow tampering, and SIP message flooding attacks. The document concludes by stating that SIP DoS attacks can render SIP services inoperable and discussing previous work on analyzing the robustness of SIP servers under DoS attacks.
The document discusses and compares Mobile IP Version 4 (MIPv4) and Mobile IP Version 6 (MIPv6), which are protocols that allow nodes to move between networks while maintaining ongoing connections. MIPv4 uses home agents and foreign agents to tunnel packets to a mobile node's care-of address, but has problems like triangular routing and security issues. MIPv6 aims to address these problems by removing the foreign agent and using other methods like return routability procedures and bindings to register locations securely.
A NEW SYSTEM ON CHIP RECONFIGURABLE GATEWAY ARCHITECTURE FOR VOICE OVER INTER...csandit
The aim of this paper is to present a new System on Chip (SoC) reconfigurable gateway
architecture for Voice over Internet Telephony (VOIP). Our motivation behind this work is
justified by the following arguments: most of VOIP solutions proposed in the market are based
on the use of a general purpose processor and a DSP circuit. In these solutions, the use of the
serial multiply accumulate circuit is very limiting for the signal processing. Also, in embedded
VOIP based DSP applications, the DSP works without MMU (memory management unit). This
is a serious limitation because VOIP solutions are multi-task based. In order to overcome these
problems, we propose a new VOIP gateway architecture built around the OpenRisc-1200-V3
processor. This last one integrates a DSP circuit as well as a MMU. The hardware architecture
is mapped into the VIRTEX-5 FPGA device. We propose a design methodology based on the
design for reuse and design with reuse concepts. We demonstrate that the proposed SoC
architecture is reconfigurable, scalable and the final RTL code can be reused for any FPGA or
ASIC technology. Performances measures, in the VIRTEX-5 FPGA device family, show that the
SOC-gateway architecture occupies 52% of the FPGA in term of slice LUT, 42% of IOBs, 60%
of bloc memory, 8% of integrated DSP, 16% of PLL and the total power is estimated at
4.3Watts.
Machine Learning applications in Voice over IPALTANAI BISHT
presented in "Women in data science Mysuru "- 2020
Media streams
Echo Cancellation
Noise Suppression
Jitter Control
Image Stabilization
Voice Activity detection
Audio fingerprinting
Echo Cancellation
Telecom Service-based Applications of ANN
Subscriber Churn and Outliers
Complains
Recharges plans
Collect CDR for daily call patterns
- identify high call volumes, or extremely long calls, or high call volumes from a particular extension
Predictive Analysis
Mean Opinion Score (MOS) - key metric for Quality of Service (QoS) of Call
predicting conversational voice quality non intrusively
Language Impact on Voice Quality assessment\
Performance
Metrics of Packet Loss on Different Codecs
VoIP provider based Applications of ANN
Anomaly detection
- Intrusion detection based on Recurrent Neural Network
(RNN) model
- Malicious System Call Sequence Detection (MSCSD)
Call Prioritization
Geographical routing
Call pattern mapping
- Bypass additional checks to remove latency
Etiquette analysis
Regulatory analysis
Telecom Fraud
Traffic Pumping
- “access stimulation” techniques to boost traffic to a high cost destination
Defraud Telecom Service Providers
- Exploitation of SIP trunks ,
- regulatory loopholes
- Premium rate numbers misused
One ring and Cut to generate Call back revenue
Blind Call Transfers
Call Cards
Vishing
VOMIT
SPIT
Detection of Fraud and Countermeasures
Call signatures
Risk Assessment
Fraud occur in off-hours
- when networks are often monitored less closely so that they can go unnoticed longer
Backpropagation Neural Network to detect SPAM calls
VoIP Intrusion Detection ( MiM)
Aggregate data from honeypot application and traffic monitoring to ANN
Recognizing attacks using ANN
Classifying Possible Intrusions
options tests; options scanning; call testing; unknown protocol; register and call; registration test, registration flooding; register attempt
Aggregate data from honeypot application and traffic monitoring to design response
ML_in_voip_altanai_wids_mysuru_sep2020
The document provides an overview of the IMS architecture from the perspective of an LTE User Equipment (UE). It describes the key components of IMS including the UE, Evolved Packet Core (EPC), IMS Core, and applications. The UE contains an ISIM and SIP User Agent. The EPC includes the PDN Gateway and PCRF. The IMS Core consists of CSCF (Proxy, Serving, Interrogating), HSS, SLF, and Media Gateways. IMS enables convergence of networks, services, and applications in an all-IP environment.
This document contains exam questions for the mobile computing course IT6601 covering topics related to mobile internet protocol and the transport layer. It includes short answer questions worth 2 marks, short note questions worth 8 marks, and longer explanation questions worth 16 marks on topics such as mobile IP, TCP, congestion control, and adapting TCP for mobile wireless networks. Sample questions address agent discovery, mobile IP tunneling and encapsulation, care-of address discovery, TCP slow start, problems using TCP in mobile networks, and approaches like indirect TCP, snooping TCP, and freeze TCP.
Performance analysis of voip traffic over integrating wireless lan and wan us...ijwmn
A simulation model is presented to analyze and evaluate the performance of VoIP based integrated
wireless LAN/WAN with taking into account various voice encoding schemes. The network model was
simulated using OPNET Modeler software. Different parameters that indicate the QoS like MOS, jitter,
end to end delay, traffic send and traffic received are calculated and analyzed in Wireless LAN/WAN
scenarios. Depending on this evaluation, Selection codecs G.729A consider the best choice for VoIP.
Rehan Rauf is seeking a career where he can apply his engineering knowledge and broaden his skills. He has a Bachelor's degree in Electronics Engineering and over 5 years of experience in network engineering roles. His experience includes work with VOIP systems, IP networks, wireless networks, and optical access networks. He has strong technical skills in Linux, networking protocols, and voice/data communications.
Mobile IP enables devices to change their Internet connection point while maintaining connectivity. It assigns a temporary IP address and uses tunneling to forward data to the device's care-of address. The Wireless Application Protocol (WAP) provides mobile access to information services over wireless networks using standards like IP, XML and HTTP. It includes the Wireless Transaction Protocol (WTP) and Wireless Transport Layer Security (WTLS) to enable secure transactions over bandwidth-limited wireless connections.
This document discusses using network coding to improve live video streaming over peer-to-peer mesh networks. It begins by introducing live video streaming and its challenges. It then discusses peer-to-peer and wireless mesh networks as infrastructures for video distribution. Network coding is presented as a technique to increase bandwidth utilization, robustness, and video quality by allowing intermediate nodes to combine packets before forwarding. The results showed that network coding can reduce delay and jitter, increase data localization, and improve bandwidth utilization and network scalability.
VoIP allows users to make voice calls over the internet instead of traditional phone lines. It works by converting voice signals to digital data packets that are transmitted over the internet and then reconverted at their destination. Key components include gateways, codecs, servers, and protocols like SIP and H.323. VoIP offers advantages like lower costs and integration with other systems but relies on internet connectivity and faces some security risks.
Comparison of DOD and OSI Model in the Internet Communicationijtsrd
The Internet protocol suite is the computer networking model and set of communications protocols used on the Internet and similar computer networks. It is commonly known as TCP IP, because it's most important protocols, the Transmission Control Protocol TCP and the Internet Protocol IP , were the first networking protocols defined in this standard. Often also called the Internet model, it was originally also known as the DoD model, because the development of the networking model was funded by DARPA, an agency of the United States Department of Defense. TCP IP provides end to end connectivity specifying how data should be packetized, addressed, transmitted, routed and received at the destination. This functionality is organized into four abstraction layers which are used to sort all related protocols according to the scope of networking involved. From lowest to highest, the layers are the link layer, containing communication technologies for a single network segment link the internet layer, connecting hosts across independent networks, thus establishing internetworking the transport layer handling host to host communication and the application layer, which provides process to process application data exchange. Our aim is describe operation and models of TCP IP suite in data communication networking. Ei Ei Khaing "Comparison of DOD and OSI Model in the Internet Communication" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-3 | Issue-5 , August 2019, URL: https://www.ijtsrd.com/papers/ijtsrd27834.pdfPaper URL: https://www.ijtsrd.com/computer-science/computer-network/27834/comparison-of-dod-and-osi-model-in-the-internet-communication/ei-ei-khaing
This document provides an overview of Abdi Kissi's internship at Ethio Telecom hosted by the Fixed Access Network department. It discusses:
1. Ethio Telecom's vision, mission, values, objectives, customers, products/services, and organizational structure.
2. Abdi's overall internship experience including objectives, how he joined the company, and an overview of the Fixed Access Network, Copper Access Network, Fiber Access Network, and Transmission divisions.
3. The benefits Abdi gained from the internship including practical experience with technologies like MSAGs, fiber splicing, troubleshooting faults, and network maintenance activities.
1. The document discusses how Multi-Protocol Label Switching (MPLS) can improve Voice over Internet Protocol (VoIP) services by enabling traffic engineering and quality of service controls.
2. MPLS allows traffic to be forwarded at layer 2 for faster routing and makes it easier to manage networks for quality of service. This helps meet the low latency and jitter requirements of real-time VoIP traffic.
3. The document analyzes VoIP performance over an MPLS network connecting three branch offices using a network monitoring tool. It finds that while MPLS provides better connectivity than the public Internet, additional steps may be needed to deliver business-quality VoIP.
Performance of MPLS-based Virtual Private Networks and Classic Virtual Privat...TELKOMNIKA JOURNAL
Multiprotocol Label Switching (MPLS) is effective in managing and utilizing available network bandwidth. It has advanced security features and a lower time delay. The existing literature has covered the performance of MPLS-based networks in relation to conventional Internet Protocol (IP) networks. But, too few literatures exist on the performance of MPLS-based Virtual Private Networks (VPN) in relation to traditional VPN networks. In this paper, a comparison is made between the effectiveness of the MPLS-VPN network and a classic VPN network using simulation studies done on OPNET®. The performance metrics used to carry out the comparison include; End to End Delay, Voice Packet Sent/Received and Label Switched Path’s Traffic. The simulation study was carried out with Voice over Internet Protocol (VoIP) as the test bed. The result of the study showed that MPLS-based VPN networks outperform classic VPN networks.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Portable voice communication system on raspberry piIRJET Journal
This document summarizes a research paper on developing a portable voice communication system using a Raspberry Pi. The system uses Asterisk software to establish communication between mobile devices over WiFi. Asterisk transforms a computer into a communications server by routing voice over internet protocol (VoIP) packets. The researchers propose setting up a Raspberry Pi and laptop on a local network to test a SIP client application. Once configured, the system would allow users to make voice or video calls between phones and laptops without a SIM card or internet by assigning IP addresses and proxies. The portable system provides benefits like low cost, remote access, and reduced wiring compared to traditional PBX phone systems.
Performance of Various Mobile IP Protocols and Security ConsiderationsCSCJournals
This document discusses and compares different mobile IP protocols. It presents an analytic model to evaluate the performance of Mobile IP (MIP), Hierarchical Mobile IP (HMIP), and Dynamic HMIP (DHMIP) based on mean signaling delay and bandwidth per call under different types of mobile terminal mobility. The model divides call holding time into small time intervals and calculates bandwidth used in each interval, accounting for both data bandwidth and signaling bandwidth during handoffs. The analysis finds that HMIP outperforms MIP and DHMIP in most cases studied due to its ability to localize registration processes and reduce signaling burden through a hierarchy of foreign agents and gateway agents.
IPTV Improvement Approach over LTE-WLAN Heterogeneous NetworksIJCNCJournal
IPTV (Internet Protocol Television) includes several video components. The IMS (IP Multimedia Subsystem) cannot differentiate between them what causes their treatment similarly. These sub-components must have different priorities because they have distinct QoS constraints. In this paper, we suggest the implementation of IPTV in a heterogeneous network that improved QoS by providing the capability to prioritize the sub traffic according to the system administrator policy. A new IPv6 flow label field definition was proposed that is ready for standardization. OPNET Modeler software is used to design our approached architecture. The results show that IPTV users receive different amounts of video data based on the stream's priority.
IPTV IMPROVEMENT APPROACH OVER LTEWLAN HETEROGENEOUS NETWORKSIJCNCJournal
IPTV (Internet Protocol Television) includes several video components. The IMS (IP Multimedia
Subsystem) cannot differentiate between them what causes their treatment similarly. These sub-components
must have different priorities because they have distinct QoS constraints. In this paper, we suggest the
implementation of IPTV in a heterogeneous network that improved QoS by providing the capability to
prioritize the sub traffic according to the system administrator policy. A new IPv6 flow label field
definition was proposed that is ready for standardization. OPNET Modeler software is used to design our
approached architecture. The results show that IPTV users receive different amounts of video data based
on the stream's priority.
This document provides a summary of Prateek's professional experience in software development for telecom and networking. Over 9.5 years, he has worked on projects involving optical networking, load balancing servers, protocol development, and customer support. His responsibilities have included technical lead roles, individual development work, design, testing, and system integration. He has strong skills in C, C++, Linux, networking protocols, data structures, and development tools like version control systems. His work experience includes roles at NEC Technology, Brocade Communication, Juniper Networks, and Huawei Technology where he contributed to projects involving network security, load balancing, network address translation, and more.
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.
This document provides an overview of Denial of Service (DoS) attacks on Session Initiation Protocol (SIP) based Voice over Internet Protocol (VoIP) infrastructure. It first introduces VoIP and SIP, describing SIP components and messages. It then discusses security issues with SIP such as eavesdropping, message tampering, and spoofing. Several types of SIP DoS attacks are classified, including SIP message payload tampering, SIP message flow tampering, and SIP message flooding attacks. The document concludes by stating that SIP DoS attacks can render SIP services inoperable and discussing previous work on analyzing the robustness of SIP servers under DoS attacks.
The document discusses and compares Mobile IP Version 4 (MIPv4) and Mobile IP Version 6 (MIPv6), which are protocols that allow nodes to move between networks while maintaining ongoing connections. MIPv4 uses home agents and foreign agents to tunnel packets to a mobile node's care-of address, but has problems like triangular routing and security issues. MIPv6 aims to address these problems by removing the foreign agent and using other methods like return routability procedures and bindings to register locations securely.
A NEW SYSTEM ON CHIP RECONFIGURABLE GATEWAY ARCHITECTURE FOR VOICE OVER INTER...csandit
The aim of this paper is to present a new System on Chip (SoC) reconfigurable gateway
architecture for Voice over Internet Telephony (VOIP). Our motivation behind this work is
justified by the following arguments: most of VOIP solutions proposed in the market are based
on the use of a general purpose processor and a DSP circuit. In these solutions, the use of the
serial multiply accumulate circuit is very limiting for the signal processing. Also, in embedded
VOIP based DSP applications, the DSP works without MMU (memory management unit). This
is a serious limitation because VOIP solutions are multi-task based. In order to overcome these
problems, we propose a new VOIP gateway architecture built around the OpenRisc-1200-V3
processor. This last one integrates a DSP circuit as well as a MMU. The hardware architecture
is mapped into the VIRTEX-5 FPGA device. We propose a design methodology based on the
design for reuse and design with reuse concepts. We demonstrate that the proposed SoC
architecture is reconfigurable, scalable and the final RTL code can be reused for any FPGA or
ASIC technology. Performances measures, in the VIRTEX-5 FPGA device family, show that the
SOC-gateway architecture occupies 52% of the FPGA in term of slice LUT, 42% of IOBs, 60%
of bloc memory, 8% of integrated DSP, 16% of PLL and the total power is estimated at
4.3Watts.
Machine Learning applications in Voice over IPALTANAI BISHT
presented in "Women in data science Mysuru "- 2020
Media streams
Echo Cancellation
Noise Suppression
Jitter Control
Image Stabilization
Voice Activity detection
Audio fingerprinting
Echo Cancellation
Telecom Service-based Applications of ANN
Subscriber Churn and Outliers
Complains
Recharges plans
Collect CDR for daily call patterns
- identify high call volumes, or extremely long calls, or high call volumes from a particular extension
Predictive Analysis
Mean Opinion Score (MOS) - key metric for Quality of Service (QoS) of Call
predicting conversational voice quality non intrusively
Language Impact on Voice Quality assessment\
Performance
Metrics of Packet Loss on Different Codecs
VoIP provider based Applications of ANN
Anomaly detection
- Intrusion detection based on Recurrent Neural Network
(RNN) model
- Malicious System Call Sequence Detection (MSCSD)
Call Prioritization
Geographical routing
Call pattern mapping
- Bypass additional checks to remove latency
Etiquette analysis
Regulatory analysis
Telecom Fraud
Traffic Pumping
- “access stimulation” techniques to boost traffic to a high cost destination
Defraud Telecom Service Providers
- Exploitation of SIP trunks ,
- regulatory loopholes
- Premium rate numbers misused
One ring and Cut to generate Call back revenue
Blind Call Transfers
Call Cards
Vishing
VOMIT
SPIT
Detection of Fraud and Countermeasures
Call signatures
Risk Assessment
Fraud occur in off-hours
- when networks are often monitored less closely so that they can go unnoticed longer
Backpropagation Neural Network to detect SPAM calls
VoIP Intrusion Detection ( MiM)
Aggregate data from honeypot application and traffic monitoring to ANN
Recognizing attacks using ANN
Classifying Possible Intrusions
options tests; options scanning; call testing; unknown protocol; register and call; registration test, registration flooding; register attempt
Aggregate data from honeypot application and traffic monitoring to design response
ML_in_voip_altanai_wids_mysuru_sep2020
The document provides an overview of the IMS architecture from the perspective of an LTE User Equipment (UE). It describes the key components of IMS including the UE, Evolved Packet Core (EPC), IMS Core, and applications. The UE contains an ISIM and SIP User Agent. The EPC includes the PDN Gateway and PCRF. The IMS Core consists of CSCF (Proxy, Serving, Interrogating), HSS, SLF, and Media Gateways. IMS enables convergence of networks, services, and applications in an all-IP environment.
This document contains exam questions for the mobile computing course IT6601 covering topics related to mobile internet protocol and the transport layer. It includes short answer questions worth 2 marks, short note questions worth 8 marks, and longer explanation questions worth 16 marks on topics such as mobile IP, TCP, congestion control, and adapting TCP for mobile wireless networks. Sample questions address agent discovery, mobile IP tunneling and encapsulation, care-of address discovery, TCP slow start, problems using TCP in mobile networks, and approaches like indirect TCP, snooping TCP, and freeze TCP.
Performance analysis of voip traffic over integrating wireless lan and wan us...ijwmn
A simulation model is presented to analyze and evaluate the performance of VoIP based integrated
wireless LAN/WAN with taking into account various voice encoding schemes. The network model was
simulated using OPNET Modeler software. Different parameters that indicate the QoS like MOS, jitter,
end to end delay, traffic send and traffic received are calculated and analyzed in Wireless LAN/WAN
scenarios. Depending on this evaluation, Selection codecs G.729A consider the best choice for VoIP.
Rehan Rauf is seeking a career where he can apply his engineering knowledge and broaden his skills. He has a Bachelor's degree in Electronics Engineering and over 5 years of experience in network engineering roles. His experience includes work with VOIP systems, IP networks, wireless networks, and optical access networks. He has strong technical skills in Linux, networking protocols, and voice/data communications.
Mobile IP enables devices to change their Internet connection point while maintaining connectivity. It assigns a temporary IP address and uses tunneling to forward data to the device's care-of address. The Wireless Application Protocol (WAP) provides mobile access to information services over wireless networks using standards like IP, XML and HTTP. It includes the Wireless Transaction Protocol (WTP) and Wireless Transport Layer Security (WTLS) to enable secure transactions over bandwidth-limited wireless connections.
This document discusses using network coding to improve live video streaming over peer-to-peer mesh networks. It begins by introducing live video streaming and its challenges. It then discusses peer-to-peer and wireless mesh networks as infrastructures for video distribution. Network coding is presented as a technique to increase bandwidth utilization, robustness, and video quality by allowing intermediate nodes to combine packets before forwarding. The results showed that network coding can reduce delay and jitter, increase data localization, and improve bandwidth utilization and network scalability.
VoIP allows users to make voice calls over the internet instead of traditional phone lines. It works by converting voice signals to digital data packets that are transmitted over the internet and then reconverted at their destination. Key components include gateways, codecs, servers, and protocols like SIP and H.323. VoIP offers advantages like lower costs and integration with other systems but relies on internet connectivity and faces some security risks.
Comparison of DOD and OSI Model in the Internet Communicationijtsrd
The Internet protocol suite is the computer networking model and set of communications protocols used on the Internet and similar computer networks. It is commonly known as TCP IP, because it's most important protocols, the Transmission Control Protocol TCP and the Internet Protocol IP , were the first networking protocols defined in this standard. Often also called the Internet model, it was originally also known as the DoD model, because the development of the networking model was funded by DARPA, an agency of the United States Department of Defense. TCP IP provides end to end connectivity specifying how data should be packetized, addressed, transmitted, routed and received at the destination. This functionality is organized into four abstraction layers which are used to sort all related protocols according to the scope of networking involved. From lowest to highest, the layers are the link layer, containing communication technologies for a single network segment link the internet layer, connecting hosts across independent networks, thus establishing internetworking the transport layer handling host to host communication and the application layer, which provides process to process application data exchange. Our aim is describe operation and models of TCP IP suite in data communication networking. Ei Ei Khaing "Comparison of DOD and OSI Model in the Internet Communication" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-3 | Issue-5 , August 2019, URL: https://www.ijtsrd.com/papers/ijtsrd27834.pdfPaper URL: https://www.ijtsrd.com/computer-science/computer-network/27834/comparison-of-dod-and-osi-model-in-the-internet-communication/ei-ei-khaing
This document provides an overview of Abdi Kissi's internship at Ethio Telecom hosted by the Fixed Access Network department. It discusses:
1. Ethio Telecom's vision, mission, values, objectives, customers, products/services, and organizational structure.
2. Abdi's overall internship experience including objectives, how he joined the company, and an overview of the Fixed Access Network, Copper Access Network, Fiber Access Network, and Transmission divisions.
3. The benefits Abdi gained from the internship including practical experience with technologies like MSAGs, fiber splicing, troubleshooting faults, and network maintenance activities.
1. The document discusses how Multi-Protocol Label Switching (MPLS) can improve Voice over Internet Protocol (VoIP) services by enabling traffic engineering and quality of service controls.
2. MPLS allows traffic to be forwarded at layer 2 for faster routing and makes it easier to manage networks for quality of service. This helps meet the low latency and jitter requirements of real-time VoIP traffic.
3. The document analyzes VoIP performance over an MPLS network connecting three branch offices using a network monitoring tool. It finds that while MPLS provides better connectivity than the public Internet, additional steps may be needed to deliver business-quality VoIP.
Performance of MPLS-based Virtual Private Networks and Classic Virtual Privat...TELKOMNIKA JOURNAL
Multiprotocol Label Switching (MPLS) is effective in managing and utilizing available network bandwidth. It has advanced security features and a lower time delay. The existing literature has covered the performance of MPLS-based networks in relation to conventional Internet Protocol (IP) networks. But, too few literatures exist on the performance of MPLS-based Virtual Private Networks (VPN) in relation to traditional VPN networks. In this paper, a comparison is made between the effectiveness of the MPLS-VPN network and a classic VPN network using simulation studies done on OPNET®. The performance metrics used to carry out the comparison include; End to End Delay, Voice Packet Sent/Received and Label Switched Path’s Traffic. The simulation study was carried out with Voice over Internet Protocol (VoIP) as the test bed. The result of the study showed that MPLS-based VPN networks outperform classic VPN networks.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Portable voice communication system on raspberry piIRJET Journal
This document summarizes a research paper on developing a portable voice communication system using a Raspberry Pi. The system uses Asterisk software to establish communication between mobile devices over WiFi. Asterisk transforms a computer into a communications server by routing voice over internet protocol (VoIP) packets. The researchers propose setting up a Raspberry Pi and laptop on a local network to test a SIP client application. Once configured, the system would allow users to make voice or video calls between phones and laptops without a SIM card or internet by assigning IP addresses and proxies. The portable system provides benefits like low cost, remote access, and reduced wiring compared to traditional PBX phone systems.
Performance of Various Mobile IP Protocols and Security ConsiderationsCSCJournals
This document discusses and compares different mobile IP protocols. It presents an analytic model to evaluate the performance of Mobile IP (MIP), Hierarchical Mobile IP (HMIP), and Dynamic HMIP (DHMIP) based on mean signaling delay and bandwidth per call under different types of mobile terminal mobility. The model divides call holding time into small time intervals and calculates bandwidth used in each interval, accounting for both data bandwidth and signaling bandwidth during handoffs. The analysis finds that HMIP outperforms MIP and DHMIP in most cases studied due to its ability to localize registration processes and reduce signaling burden through a hierarchy of foreign agents and gateway agents.
IPTV Improvement Approach over LTE-WLAN Heterogeneous NetworksIJCNCJournal
IPTV (Internet Protocol Television) includes several video components. The IMS (IP Multimedia Subsystem) cannot differentiate between them what causes their treatment similarly. These sub-components must have different priorities because they have distinct QoS constraints. In this paper, we suggest the implementation of IPTV in a heterogeneous network that improved QoS by providing the capability to prioritize the sub traffic according to the system administrator policy. A new IPv6 flow label field definition was proposed that is ready for standardization. OPNET Modeler software is used to design our approached architecture. The results show that IPTV users receive different amounts of video data based on the stream's priority.
IPTV IMPROVEMENT APPROACH OVER LTEWLAN HETEROGENEOUS NETWORKSIJCNCJournal
IPTV (Internet Protocol Television) includes several video components. The IMS (IP Multimedia
Subsystem) cannot differentiate between them what causes their treatment similarly. These sub-components
must have different priorities because they have distinct QoS constraints. In this paper, we suggest the
implementation of IPTV in a heterogeneous network that improved QoS by providing the capability to
prioritize the sub traffic according to the system administrator policy. A new IPv6 flow label field
definition was proposed that is ready for standardization. OPNET Modeler software is used to design our
approached architecture. The results show that IPTV users receive different amounts of video data based
on the stream's priority.
Delay Efficient Method for Delivering IPTV ServicesIJERA Editor
Internet Protocol Television (IPTV) is a system through which Internet television services are delivered using
the architecture and networking methods of the Internet Protocol Suite over a packet-switched network
infrastructure, e.g., the Internet and broadband Internet access networks, instead of being delivered through
traditional radio frequency broadcast, satellite signal, and cable television (CATV) formats. IPTV provides
mainly three services: live TV, catch up TV, and video on demand (VoD).This paper focuses on delivering the
live TV services by exploiting the virtualised cloud architecture of the IPTV and statistical multiplexing. The
VoD tasks are prescheduled so that there will be less Instant Channel Change (ICC) delay. We select a proper
scheduling algorithm for rescheduling the VoD tasks. We then implement the scheduling algorithm for preshifting
the VoD tasks.
An SDN Based Approach To Measuring And Optimizing ABR Video Quality Of Experi...Cisco Service Provider
Reprinted with permission of NCTA, from the 2014 Cable Connection Spring Technical Forum Conference Proceedings. For more information on Cisco video solutions, visit: http://www.cisco.com/c/en/us/products/video/index.html
Performance Evaluation of Interactive Video Streaming over WiMAX Network IJECEIAES
Nowadays, the desire of internet access and the need of digital encodings have influenced quite a large number of users to access high quality video application. Offering multimedia services not only to the wired but to wireless mobile client is becoming more viable. In wireless medium, videostreaming still has high resource requirements, for example, bandwidth, traffic priority, smooth play-backs. Therefore, bandwidth demands of these applications are far exceeding the capacity of 3G and Wireless Local Area Networks (LANs). The current research demonstrates the introductory understanding of the Worldwide Interoperability for Microwave Access (WiMax) network, applications, the mechanisms, its potential features, and techniques used to provide QoS in WiMAX, and lastly the network is simulated to report the diverse requirements of streamed video conferencing traffic and its specifications. For this purpose two input parameters of video traffic are selected, i.e, refresh rate, which is monitored in terms of frames per second and pixel resolutions which basically counts the number of pixels in digital imaging. The network model is developed in OPNET. Different outcomes from simulation based models are analyzed and appropriate reasons are also discussed. Apart from this, the second aim of the current research is to address whether WiMAX access technology for streaming video applications could provide comparable network performance to Asymmetric Digital Subscriber Line (ADSL). For this purpose network metrices such as End to End delay and throughput is taken into consideration for optimization.
This document provides an application progress report for research on evaluating the performance of LTE-Advanced (LTE-A) networks for quality of service (QoS) in internet protocol television (IPTV) from June 2019 to December 2019. The researcher introduces LTE and QoS, describes the LTE architecture, and states the problem of ensuring best video quality for users. Observations from simulations on cell data and extracted data are summarized. Next steps include writing a research paper.
This document summarizes a research paper that proposes a framework to improve quality of experience and energy efficiency for heterogeneous wireless multimedia broadcast receivers. The framework groups users based on their device capabilities and channel conditions. It broadcasts scalable video streams that are encoded with different layers to support different groups. Time slicing is used to allow discontinuous reception and energy savings by turning radios off between bursts. A game theoretic model is used to optimize source encoding, transmission scheduling, and modulation/coding to maximize reception quality and network capacity while balancing energy usage. Evaluation shows the approach enables 75-95% energy savings.
Triple Play service is a marketing term for provisioning of two bandwidth-intensive services, high-speed Internet access and television, and a less bandwidth-demanding (but more latency-sensitive) telephone service, over a single broadband connection.
In this thesis, the effect of mobility of mobile WiMAX subscribers on video on demand (VOD) over WiMAX is analyzed by considering the scalable video coding (SVC) codes for video streaming. This experiment has been carried out using OPNET modeller 14.5. To compare the performance of Internet Protocol television (IPTV) over WiMAX, the packet delay variation, packet end to end delay, delay and load matrices are used. The result shows that after certain speed, the load increases, the delay again decreases and there is no change in the packet delay variation and packet end to end delay.
Key words: WiMAX, OPNET, scalable video coding (SVC), wireless networks, IEEE 802.16, internet protocol television (IPTV).
MODELLING AND PREFORMANCE ANALYSIS FOR VIDEO ON DEMAND PRIOR STORING SERVER ijwmn
The document proposes a new architecture for video-on-demand (VoD) in 4G LTE networks that includes prior storage servers in each eNodeB base station. The prior storage server is divided into two levels - Prior Storage 1 and Prior Storage 2 - to store video content segments based on their popularity and utility. A partial prior storage strategy is used to avoid replacing popular video content with unpopular content. The performance of the proposed LTE network architecture is modeled and analyzed using RT-PEPA (Real-Time Performance Evaluation Process Algebra) and simulation results show improvements in quality of service parameters like packet loss, delay, and jitter.
Optimal Rate Allocation and Lost Packet Retransmission in Video StreamingIRJET Journal
This document summarizes research on optimal rate allocation and lost packet retransmission for video streaming over wireless networks. It discusses challenges including calculating desired transmission rates based on network conditions, scaling video output rates, and differentiating between packet loss due to congestion versus link errors. A block diagram is presented showing the transmission system, including a link adaptation scheme to adjust transmission parameters based on channel feedback. Formulas are provided for an affine function used in the rate allocation algorithm. Finally, graphs are proposed to evaluate the packet delivery ratio achieved by different streaming approaches.
Despite the increasing roll-out of broadband terrestrial services like DSL, a
significant amount of households worldwide are deprived from fast broadband
access services. Bridging this digital divide is high on the agenda of decision
makers because broadband penetration has high economic impact on a
country. The service cannot be limited to cities. This requirement is translated
into 100% service obligations for Internet Service Providers (ISPs) covering
the whole territory of a country or region. Running these services in an
economically viable way is a major challenge.
This document discusses IPTV (Internet Protocol Television) which delivers television programming over private IP networks. Some key points:
- IPTV provides a substitute for traditional cable/satellite TV by delivering hundreds of channels via set-top boxes connected to televisions.
- Service providers choose IPTV as it allows delivering voice, high-speed data, and video over a single network platform.
- IPTV networks privately deliver continuous video streams simultaneously to many viewers. Set-top boxes receive streams and display them on televisions.
QoE-enabled big video streaming for large-scale heterogeneous clients and net...redpel dot com
This document summarizes a research paper on providing quality of experience (QoE)-enabled video streaming for heterogeneous clients and networks in smart cities. It discusses the growth of video traffic and challenges in ensuring high QoE across different devices. The paper reviews video broadcasting technologies, coding methods for scalability and flexibility, and presents a paradigm for QoE-mapped joint coding and cross-layer transmission to dynamically adapt to different devices and networks. It evaluates system performance in terms of broadcasting efficiency and discusses open areas for future research.
This paper develops neural network models that can predict user quality of experience (QoE) for Internet Protocol television (IPTV) applications in real time based on network measurements. The models account for multiple video resolutions, audio/video codecs, and network conditions including jitter, packet loss, and router queuing disciplines. The models were trained using data from objective network simulations and subjective human experiments evaluating mean opinion scores for quality. Evaluation shows the models accurately and quickly predict user QoE for IPTV under different conditions and can be used to monitor network quality in real-time.
This document summarizes a survey paper on virtualized cloud-based IPTV systems. It discusses how IPTV delivery over the internet places high demands on service provider resources. A virtualized cloud infrastructure allows resources to be shared and dynamically allocated to better handle instant channel change workload bursts and anticipate changes. The paper formulates an optimization problem to determine the optimal number of servers needed at different times to support live TV and video-on-demand services while minimizing costs. Various cost functions are analyzed. Benefits of cloud-based IPTV include interactivity, video on demand, ease of use, and integration of services.
With the Intel® Ethernet 10 Gigabit Converged Network Adapter X520, Qing Niu builds a new generation of CDN streaming servers to increase its concurrency by three times, delivering benefits for its customers. The new servers use the adapter's RSS and NUMA architecture to improve throughput performance, balancing loads across multiple processor cores. This allows the servers to achieve streaming throughput of up to 18Gbps while serving over 20,000 concurrent users efficiently.
This document evaluates the performance of IPTV video streaming over WiMAX networks under different terrain environments, including free space, outdoor to indoor, and pedestrian environments. It uses OPNET simulations to analyze network statistics such as packet loss, path loss, delay, and throughput. The results show that free space terrain has the lowest path loss and packet delay, while outdoor to indoor and pedestrian environments have higher path loss and delay. Specifically, free space path loss was around 100dB while outdoor environments was around 145dB. Additionally, packet loss was highest for outdoor scenarios due to lower signal to noise ratios in those environments. In general, more obstructed environments led to worse performance for IPTV video streaming over WiMAX networks.
Internet Protocol Television (IPTV) is a system that delivers digital television services over a managed broadband connection rather than traditional cable or satellite. IPTV uses Internet Protocol over a broadband connection to deliver digital television programming, movies on demand, and other interactive multimedia services. It provides a more personalized television experience than traditional cable or satellite through features like an interactive program guide, picture-in-picture functionality, and the ability to access photos or music from a personal computer on the television. IPTV is expected to grow significantly as broadband access expands globally. Major telecommunications providers are exploring IPTV as a new revenue opportunity and as a defensive measure against competition from cable television services.
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology
Qo s management for mobile satellite communicationeSAT Journals
Abstract In this paper, a cross-layer architecture (QoSatAr) is developed to provide end-to-end quality of service (QoS) guarantees for Internet protocol (IP) traffic over the Digital Video Broadcasting-Second generation (DVB-S2) satellite systems. The architecture design is based on a cross-layer optimization between the physical layer and the network layer to provide QoS provisioning based on the bandwidth availability present in the DVB-S2 satellite channel. One of the most important aspects of the architecture design is that QoSatAr is able to guarantee the QoS requirements for specific traffic flows considering a single parameter: the bandwidth availability which is set at the physical layer (considering adaptive code and modulation adaptation) and sent to the network layer by means of a cross-layer optimization. The architecture has been evaluated using the NS-2 simulator. Keywords: QoSatAr, DVB-S2, ACM, RQM, DiffServ
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In his public lecture, Christian Timmerer provides insights into the fascinating history of video streaming, starting from its humble beginnings before YouTube to the groundbreaking technologies that now dominate platforms like Netflix and ORF ON. Timmerer also presents provocative contributions of his own that have significantly influenced the industry. He concludes by looking at future challenges and invites the audience to join in a discussion.
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Discover how Standard Chartered Bank harnessed the power of Neo4j to transform complex data access challenges into a dynamic, scalable graph database solution. This keynote will cover their journey from initial adoption to deploying a fully automated, enterprise-grade causal cluster, highlighting key strategies for modelling organisational changes and ensuring robust disaster recovery. Learn how these innovations have not only enhanced Standard Chartered Bank’s data infrastructure but also positioned them as pioneers in the banking sector’s adoption of graph technology.
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The Optimization of IPTV Service Through SDN In A MEC Architecture, Respective Simulations In OMNeT ++
1. Esmeralda Hysenbelliu & Arjan Durresi
International Journal of Software Engineering (IJSE), Volume (9) : Issue (3) : 2021 63
ISSN: 2180-1320, https://www.cscjournals.org/journals/IJSE/description.php
The Optimization of IPTV Service Through SDN In A MEC
Architecture, Respective Simulations In OMNeT ++
Esmeralda Hysenbelliu esmeralda.hysenbelliu@gmail.com
Faculty of Information Technology
Computer Engineering Department
Polytechnic University, Tirana, Albania
Arjan Durresi adurresi@iupui.edu
Computer and Information Science Dept
Indiana University Purdue,
IN 46032, USA
Abstract
The aim of this paper is to present the ‘Power’ of SDN Technology and MEC Technic in
improving the delivering of IPTV Service. Those days, the IPTV end –users are tremendous
increased all over the world , but in the same time also the complains for receiving these prepaid
real time multimedial services like; high latency, high bandwidth, low performance and low
QoE/QoS. On the other end, IPTV Distributors need a new system, technics, network solutions to
distribute content continuesly and simultaneously to all active end-users with high-quality, low-
latency and high Performance, thus monitoring and re-configuring this ‘Big Data’ require high
Bandwidth by causing difficult problems by offering it affecting in the same time the price and
QoE/QoSperformance of delivered service.
For this reason, we have achieved to optimize the IPTV service by applying SDN solution in a
MEC Architecture (Multiple-Access Edge Computing). In this way , through MEC Technology and
SDN, it is possible to receive an IPTV service with Low Latency, High Performance and Low
Bandwidth by solving successfully all the problems faced by the actual IPTV Operators. These
improvements of delivering IPTV service through MEC will be demonstrated by using the OMNet
+++ simulator in an LTE-A mobile network. The results show clearly that by applying the MEC
technique in the LTE-A network for receiving IPTV Service through SDN Network, the service
was delivered with latency decreased by >90% (compared to the cases when the MEC technique
is not applied), with PacketLoss of almost 0 and with high performance QoE. In addition these
strong Contributions, the ‘Big’ innovation achieved in this work through simulations is that the
quality of delivered IPTV Service did not change according to the increasing of the end-users.This
latency of delivering the video streaming services did not change. This means that the IPTV
Service providers will increase their benefits by ensuring in the same time also the delivering of
service with high quality and performance toward innumerous end users. Consequently, MEC
Technology and SDN solution will be the two right and "smart" network choices that will boost the
development of the 5th Mobile generation and will significantly improve the benefit of Video
Streaming services offered by current providers worldwide (Netflix, HULU, Amazon Prime,
YouTube, etc).
Keywords: SDN, MEC, IPTV, Latency.
1. INTRODUCTION
Nowadays, the end users’ demands for Multimedia services like TV, Video, Audio, Text, Graphics
and Data have increased tremendously. Most researchers believe that IP-based TV service, also
known as IPTV service, is a key opportunity for operators around the world to increase their
profits by offering video over IP networks. The IPTV service itself supports QoS (Quality of
2. Esmeralda Hysenbelliu & Arjan Durresi
International Journal of Software Engineering (IJSE), Volume (9) : Issue (3) : 2021 64
ISSN: 2180-1320, https://www.cscjournals.org/journals/IJSE/description.php
Service), QoE(Quality of Experience or Perception made by end users), Security, Interoperability
and the Level of Reliability required by IP networks. Since IPTV service is a paid service to
receive all live broadcast TV channels, this service providers are focused on providing it in real
time with very high QoE and low cost. To achieve this goal, IPTV Operators need a system to
distribute content to high-quality, low-latency video subscribers located in different geographical
locations. On the other hand, IPTV end users are expecting to get the service inhigh quality, low
cost and low latency. To ensure the provision of IPTV service in accordance with the
requirements of end users, providers require high bandwidth and an efficient way of sending
video content from the source to end users.
In addition, typical IPTV networks have to deliver video content continuously and simultaneously
to thousands of end users, thus monitoring and re-configuring these systems poses a very
difficult problem. Multimedia services, which were previously provided by network operators and
on a dedicated network, have now migrated to an open Internet. This way of gaining service is
called OTT (Over-The-Top). OTT systems(Nam et al ., 2016) such as 'Nettflix', 'Youtube' and
'Hulu' for video service delivery can be challenging because end users, IPTV service providers
and network operators (ISPs) do not have an overview of network point-to-point conditions. In this
case, IPTV service providers do not have the ability to simultaneously realize the change of ISPs
and the logic of that small part of the network that currently covers the user. In addition to that,
the content server, once it starts broadcasting a channel, very rarely switches to another node.
Consequently, due to unstable network conditions, end users often encounter re-buffering to the
end of a video and in turn, the obtained service quality is low. That is how the idea for this
research came up, the aim of which is to solve all the above problems using a new network
technology, the SDN (Software Defined Networking) one. (Sezer et al., 2016).
SDN technology (Hysenbelliu et al ., 2015) is an ideal solution for managing IPTV networks
because it has the potential to detect video quality problems within the network core with its new
sizes and mechanisms. This 'Smart Solution' called SDN, supports Key Performance Indicators
(KPIs) and contains powerful network monitoring and reconfiguration features. Through it, IPTV
service providers may choose the best content server when an end user requiresto receive video
service, enabling the optimization of obtaining IPTV service and monitoring of network conditions.
Furthermore, the implementation of SDN technology for obtaining IPTV service reduces the
OpEx/CapEx provision costs and provides an improved service (Hasan et al., 2020). Unlike
traditional networks, the SDN network:
● separates the data plan that passes full-speed traffic from the control plan, which makes
decisions about how to pass scalable traffic over a long period of time
● already provides a well-defined interface between the shared control plan and that of the
data, including a set of abstractions for network devices that hide many details within
them
● migrates the control plan logic to a centralized logic controller (SDN Controller) that
employs the global view of network resources and applications requirements ‘knowledge
to create and optimize general rules.
2. IMPLEMENTATION OF SDN TECHNOLOGY IN SMC IPTV ISP DATA
CENTER
In this section, we have implemented SDN Technology in the real IPTV service provided, called
‘SMC IPTV ISP’, which is always interested in the continuous improvement of IPTV service (
Esmeralda et al ., 2017-Esmeralda et al., 2018). Following the implementation of the SDN
technology through Virtualization, the QoEof received IPTV Service, from end users’ point of
view, increased significantly, the cost of receiving IPTV service decreased, and through the
implementation of the GPU processor (Esmeralda et al., 2017) the service performance
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increased as well. The demand for such services has increased terribly, thus in turn increasing at
the same time the bandwidth required in the network, the service delivery delays, as well as
difficulties in the management and check of enormous data. In addition to that, the increase of IoT
users, 5G services, mobile and social media users has led to an increase in the number of
devices on the network as well as an increase in the central cloud traffic (increases the load and
computing capabilities). This causes problems for real-time applications where latency is a very
important factor. This introduces MEC (ETSI et al., 2016, Sabella at al., 2016) technology, which
shifts resources across network boundaries and close to mobile and IoT users by providing real-
time video streaming services with low latency and bandwidth, as well as high performance.
Based on the principles of SDN technology, SMC IPTV ISP has built an optimized network
architecture for providing low latency and high QoE IPTV service. This architecture is based on
SDN and MEC technologies where MEC technology is made of three layers: the MEC
Application, the MEC Platform itself and the Abstract layer.
Abstraction
MEC Platform
MEC Aplication
Service Aplication
Video
Internet
IPTV
Base Services
SMC
IPTV ISP
Switch OF/RAN
MEC
Video
Internet
IPTV
I1
I1
I3
API
Edge Network
FIGURE 1: IPTV Service Optimization Architecture according to MEC based on SDN.
As seen in Fig 1, a copy of the services provided by the SMC IPTV ISP data center has been
transferred to the end-user boundaries, i.e., to the MEC Platform. The aim is to receive an IPTV
service with less delays and low latency, as well as high QoE and performance. Also, in this
architecture, there are appeared two ways of receiving the IPTV service:
a) The first way is the traditional way (in red) where the end user wants to watch a real time TV
channel or a streaming video by making the internet request to the real content server owned by
the SMC IPTV service provider ISP. Whereas, the Routing part of the requests coming to the
network is realized according to the principles of the SDN solution provided by the OpenFlow
Switch.
b) The second way is the blue one, where two mobile users perform video streaming via HTTP as
a specific MEC application. At the moment of starting video streaming, the above architecture has
the possibility to program the Routing patches in such a way that the Video application is
downloaded from the MEC Server and not from the real server of the SMC IPTV ISP Internet
service provider. The MEC network also adjusts the streaming speed according to RAN status.
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Consequently, the end user’s deliver the optimized und enhanced streaming services like Video,
IPTV with low delay and high QoE.
3. THREE SCENARIO SIMULATIONS WITH OMNET++ SIMULATOR
The main purpose of this part is to simplify the demonstration of receiving the already optimized
and improved IPTV service using the SDN Network Solution and MEC Technology principles.
Such simulations will be performed through theOMNeT ++ simulator platform. This simulator
uses the NED language (Network Element Description language) to define the network structure
and the incoming or outgoing data is placed in the '.ini' file (commonly called omnetpp.ini),
supports all types of Wireless, and wired protocols.
As a first scenario it will be simulated the benefit of IPTV service from an end user via the
internet using real time streaming protocols like RTP and RTCP.
As a second scenario it will be simulated the benefit of IPTV streaming service by applying the
SDN solution and using the ‘OpenFlow’ controller.
Finally, as a third simulation scenario it will be realized the benefit of IPTV service through the
implementation of MEC Technology based on an LTE-A network.
Some of the most important parameters of the transmission link for multimedia services such as
Video, Audio, IPTV, etc. in real time are point-to-point end delays, Bit Error Rate (BER), service
benefit latency, Jitter, SNIR and PacketLoss.
3.1 Scenario 1: Receiving IPTV service through internet using real-time streaming
protocols such as RTP and RTCP
These are the manuscript preparation guidelines used as a standard template. Author must follow
Simulation of video streaming data transmission (or IPTV service) uses the RTP protocol in the
INET framework. Specifically, our simulation will be performed on the OMNeT ++ platform,
version 4.2.2 and with the INET 2.2 framework. In the simulation that we will perform for the
transmission of the Video service according to the H.264 coding standard, the following steps
need to be performed:
1. The network is created in the INET framework of OMNeT ++, and specifically it is called
'Sk1IPTV'.
2. In this step, there are created the sending and receiving modules, namely the specific
RTP hosts called 'Sender' and ‘Receiver’, which are responsible for sending, and
receiving H.264 video data. These two modules complete the packaging and
disassembly process.
3. It will be created video data routing elements, namely 3 IPv4 Routers that support
wireless, PPP, Ethernet and external interfaces
4. Fourthly, the submodule connections are created with each other, which in this case are
Ethernet connections as shown in Figure 2.
In this way, the network topology is with success created and is ready for simulation via the
Sk1IPTV.ned file, as well as also the parameters of the ‘Sender’, ‘Receiver’ modules and other
submodules by successfully generating the Sk1IPTV.ini file. Communication has also been
logically established between the 'Receiving and Sending' modules, which will exchange IPTV
video streaming with each other.
The simulation process appears through a special interface called Tkenv, which shows in detail
the specific events, time, number of actions of modules and submodules, etc. This interface also
verifies the validity of the network topology, which visually shows in real time, which modules
receive or send data packets, as it is show in Figure 3.
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In this first simulation scenario, we will consider the evaluation of some very important
parameters that directly affect the quality of images of IPTV service received by end users; the
evaluation of the Point-to-point delay of delivering the video streaming service, Queue packet
time and Packet loss. The main purpose of these simulations is to highlight the improvement of
the quality of IPTV service from the implementation of new network solutions such as SDN and
MEC technology.
FIGURE 2: Transmission of H.264 video data between a Sender and a receiver (.ned file).
FIGURE 3: The final result in the Tkenv interface.
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a) Estimation of the Point to Point delay
According to the network topology created in fig. 2, the network is configured in the .ned file with
the corresponding modules and submodules, as well as their parameters are defined by creating
the .ini file
The first simulation is performed for 300s and the Video sent by the Sending Module is
lego_video.mpg.gdf, with a size of 54,528 Bytes and Bandwidth = 10Mbps. Final simulation data
generation in OMNeT ++ IDE is realized in three forms; vector data, scalar data and histogram
data. Then, the final delay between 'Sender' and Recipient while obtaining video streaming is as
in the figure below:
FIGURE 4: Estimation of point-to-point delay in delivering IPTV service.
The final delay is received in the RTP Receiving Module, which reaches a maximum value of 54
seconds, in the bins interval (0.00225804484, 0.0024194981733333) and no delay is displayed
from the RTP Sender module.
b) Estimation of packet waiting time for transmission
As shown in Figure 5, the longest packet waiting time occurred on Router 1, and specifically 93
seconds in the bins interval (1.346E-5 .. 1.346E-5.)
In order to estimate the delivering time of packets during the transmission of video streaming from
the Sender module to the Receiver Module, we will take again in consideration the ‘Histogram
data’.
FIGURE 5: Estimation of the packets’queuing delay.
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c) Estimation of Packet Loss
One of the most important parameters during transmission of video streaming is also that of
PacketLoss, which the smaller it is, the better the quality of service and performance is delivered
by the end users.To get an accurate definition of this parameter, we have estimated the number
of packets sent from the udp Sender Module to the rtp Receiver Module (in bytes), as well as the
number of packets received from the rtp Receiver Module (in bytes). Logically their difference
determines the value of packet loss. From the generated conclusions file, we considered the
scalar data. As shown in the figure below, the number of packets sent by the udp sender is 55802
Bytes,while the number of packets received by ‘Receiver rtp’ is 53410 Bytes. Consequently, the
number of lost packets is 55802 - 53410 = 2392 packets. Referring to the following formula for
calculating packet loss in percentage, then for the first simulation scenario, this packet loss P (H)
is 4.28%.
FIGURE 6: Evaluation of packets sent by the Sender module and that ones received from the Receiver
Module.
P (H) = P (h) / N
Where P (h) is the number of packets lost
N is the total number of packages sent by the Sender
3.2 Scenario 2: Receiving from IPTV Service using SDN Solution
In the OMNeT 4.2.2 simulator that we will use to simulate the delivering of video streaming
service using the SDN solution, we need to integrate OpenFlow components. OpenFlow assists
in the development and use of SDN technology by providing high flexibility in routing network
leaks, as well as allowing you to change the behavior of a portion of the network without affecting
whole data traffic. This is achieved by separating the control plan in the OpenFlow Switches
network from the data plan. In this way, using the INET framework, OpenFlow Switches and a
Controller, we have managed to build the network topology in OMNet. The controller is
responsible for all streaming data routing decisions as well as changing packet-forwarding rules
in Switch. To realize the simulation of receiving the IPTV service on an SDN network, we have
used a Controller, a Server that carries video data according to the RTP protocol configurations,
two OpenFlow Switches and end users who will receive IPTV streaming service; Client1 and
Client2. The steps followed for building the network topology in NED language and all the other
actions are same as in the first scenario, in thus a way the NED topology is SDN1.ned as it is
showed in Figure 7.
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In the simulation that we will perform we have taken as a behavior of the Controller that of
'Forwarding' because the Controller module itself has complete knowledge of the whole network,
thus realizing the sending of packets to the OpenFlowSwitch_it, which is positioned in the path of
required for data transmission to the end user.
The simulation is presented through the Tkenv interface, which shows in detail the specific
events, time, number of actions of modules and submodules, etc. This interface, as shown in
Figure 8, shows in detail which modules and submodules in the SDN1.ned network topology
exchange information with each other.
FIGURE 7: Building the SDN network topology according to the NED language (SDN1.ned).
The initial parameters considered for the SDN1 network are: Bandwidth 100 Mbps, delay is
0.000001s, Clients are standard Hosts, Throughtput 10µs, Server is an RTP Host and the
uploaded video is the same as in the first scenario; lego_video.mpg.gdf with size 54528 Bytes.
This video is encoded in H.264 format. The simulation time is 600 s.
FIGURE 8: The final result in the Tkenv interface.
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The parameters we will evaluate in this second scenario are:
a) Average RTT (Round Trip Time)
This time, called RTT is the same as end to end delay. This parameter estimates the time it takes
for packets to be transmitted from a given ‘Sender’ to a ‘Receiver’. The smaller this delay, the
better the quality of service that the Customer perceives (So the QoE increases). SDN solution
has high expectations to enable the delivering of Video Streaming service with small RTT.
FIGURE 9: Estimation of RTT average time for OpenFlow Controller and Switches.
As seen in Figure 9, the biggest delay is shown by the OpenFlow Switch, which goes specifically
to 0.332894725869979 s.
Compared to the point-to-point delay estimated from the first scenario (54 s) without
applying the SDN solution, this delay is reduced by 61.62%, which increases direct the
quality of service delivered from the end user's (QoE).
b) RTO (Recovery time Objective)
RTO time is a paramter that measures the time that data is invalid or inaccessible to the network.
Mostly this time will be estimated at SDN Controller and OpenFlow Switch at TCP level. Similar to
PaketLoss in the first simulation scenario, this time should be as short as possible in order to
deliver with high-performance the real-time streaming services like IPTV. As you can see the
result in the figure below, this time for the OpenFlow Controller and Switches is almost the same,
specifically it achieves the value 0.65625s.
FIGURE 10: RTO estimation for OpenFlow Controller and Switches.
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c) Estimation of PacketLoss
To estimate the number of packet’s Bytes being lost (Packetloss), specifically to Client1 and
Client2 in this time interval, we will calculate the number of bytes of packets sent by the RTP
Server and those that actually have been successfully received by end users. As shown in Figure
11, the number of bytes of packets sent by the Video Streaming Server is 2432.
While the number of packet’s bytes successfully received by Client1 and Client2 are:
Client1 has successfully received 2368 bytes of packets
Client2 has successfully received 2432 bytes of packets
FIGURE 11: The number of packets’ bytes sent by the Video streaming Server.
FIGURE 12: The number of packets’ bytes successfully delivered by Client1 and Client2.
As shown in the Figure 12, Client2 has successfully received 2432 packets’ bytes, thus in Client2
there is no PacketLoss. Referring to the formula for calculating packet loss (First Scenario;
Estimation of packet loss) in percentage, then packet loss in percentage referred to Client1 is: P
(H) = 2.631%.
Comparing this value with the value of packet loss from Scenario 1 (P (H) = 4.28%) where
delivering IPTV streaming service is realized without applying the SDN solution, then we
conclude that IPTV streaming service through SDN solution is delivered with a relatively
low PacketLoss, decreasing by 61%.
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3.3 Scenario 3: Delivering IPTV Service using MEC Technology
Recently, the demands for Video streaming services have increased significantly. Powerful
service providers such as ‘Netflix’, ‘Hulu’, ‘Amazon Prime Video’, etc. are facing major problems
with network capacity, fast bandwidth consumption, large amount of data management, resource
efficiency and saving of required QoS service quality. Requests for Video Streaming services
have become a necessity not only for Wireless end users, but also for mobile ones.
Consequently, the delivering of these services in real time needs to be provided with low Latency
and high QoE. Last year, there were reported numerous problems with the delay in getting
streaming services from Netflix service users. In this way, it is necessary to realize the placement
of resources of these services (eg Content) and the realization of data processing and
management as close as possible to the end user, finally at the age of the network, in order to
reduce delays, jitter and increase the quality of service QoE. MEC technology enables the
deployment of a computing platform in the Cloud (called MEC Host) as close as possible to
mobile users on 4G and 5G networks in order to receive the service with Low Latency and high
performance. In this way, for this third part of the simulations we have chosen to realize the
benefit of the Video streaming service using MEC technology in an LTE-A mobile network. The
main purpose is to evaluate the benefit of delivering IPTV video streaming service with low
latency using MEC technology combined with SDN and NVF network solutions. To realize this
simulation scenario we will use the simulator OMNeT ++, version 5.1.1. The MEC architecture will
be integrated within the LTE-A network through the SimuLTE framework (Giovanni et al., 2016)
enabling the evaluation of the performance of MEC services according to the real conditions of a
network infrastructure. The main reason why we chose the LTE network for the implementation of
MEC technology and to finalize the delivering of video streaming with very low latency, is
because the MEC in interaction with the developed standards of LTE technology will be a by very
important component for building 5G mobile network.
To build the MEC Host model in the SimuLTE framework integrated into OMNet, we have
referred to the work done by the authors in (Giovanni et al., 2018). As shown in the figure below,
Host MEC is built as a composite module, which has within it four sub-modules; Simple ME
Platform Module, Simple ME Application Module, Video Streaming Server Module and PFGTP
Endpoint Module.
FIGURE 13: Building MEC Host in SimuLTE.
The ME Application Module is the module that receives video streaming requests from EU end
user’s, which go first to the video streaming udp server via the PFGTP endpoint. After that, the
Video streaming server will communicate with the base radio station called eNB, which receives
requests for video streaming from the respective EU. The GTP module can be placed inside the
EPC part of the LTE network and communication between them is tunneled via the GTP protocol.
This module performs the encapsulation and de-capsulation of data packets within the GTP
package.
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In this way, we will build the LTE network in OMENeT by placing the MEC Host above directly on
the eNB base station as shown in the figure below.
FIGURE 14: Construction of LTE network topology in OMNeT based on MEC Host connection directly to
eNB base station.
In this way, the delivering of video streaming service from the EU could be realized with low
latency, very low packet loss and high performance.
We have built the .ned file and .ini configuration file. In the first case of the simulation, we will get
the same 5MiB video size (5.24288 MB), with packet length of 500B, packet delivery interval of
10ms and 20 EU.
The simulation results are in .vec and .sca files.
In general, the concept of "Latency" indicates the delay between a source and a certain
destination caused by a certain network (which can be a mobile network, a wireless network,
etc.). Since this parameter is a very important aspect in the development of the 5G mobile
generation where it is expected that this parameter called ‘latency’ will be much lower compared
to the previous 4G generation, it is really necessary to determine exactly what this delay is and
where it depends. Thus, the latency classification in the network referring to 3GPPis divided into;
• Latency according to the control plan - which includes the delay of the transition of end users
from the state of 'Quiet' to the 'Active' in both, the Radio network and the core network.
• User plan latency - is the time it takes for a valid packet in the IP layer of the RAN node or EU
end user to transfer and be valid again in the RAN or EU. As explained above, the RAN node is
the node that provides the Radio access interface to the Core network.
In our simulation case, we will evaluate the Latency according to the user plan which constitutes
the time it takes for a packet from the moment it becomes valid at the source and until it is valid at
the destination node.
As seen in the Figure below, the latency of delivering video service from 20 EU according to MEC
technology is 0.004 s.
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FIGURE 15: Latency of delivering Video Streaming service based on MEC technology for 20 UE.
Comparing it to the result of two other scenarios, where we have not applied MEC
technology, Latency has been decreased by >90% (0.332s based to scenario 2 and by
53,996s based to scenario 1).
From the analysis of the simulation results files, we noticed that in the case of applying the MEC
technology for the delivering of the Video Streaming service, the packet loss is 0. This means that
the QoE quality of the service perceived by the end users is very high.
As seen in Figure 16, with the increase in the number of End Users, specifically from 20 to 100
UE, and from 100 to 1000 UE with unchanged packet length of 500 B, unmodified packet delivery
interval 10ms, Service benefit latency video streaming is again approximately 0.004, therefore it
does not change. In addition to that, PacketLoss is 0.
These results show that MEC Technology and SDN will be the two right and "smart" network
choices that will boost the development of the 5th Mobile generation and will significantly improve
the benefit of Video Streaming services offered by current providers worldwide (Netflix, HULU or
Amazon Prime Video).
FIGURE 16: Comparison of the Latency of delivering Video Streaming service based on MEC technology for
20 UE and 100 UE.
4. CONCLUSION
This paper addresses the improvement and optimization of delivering IPTV service through the
application of SDN solution and network virtualization. We have also built an optimized IPTV
service delivery architecture based on MEC and SDN technology, in which streaming data was
moved to small clouds as close as possible to the end users. Consequently, video streaming
services like IPTV, VoDetcwere delivered with low Latency and Bandwidth, and in the same time
with High Performance QoE.
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To evaluate the optimizations of IPTV service based on the SDN network solution and according
to the MEC technique, we have built three simulation scenarios in OMNet ++, through which we
have obtained the following results:
1. The final RTT delay of delivering IPTV service via SDN, compared to delivering the IPTV
service through RTP and RTCP streaming protocols has been reduced by 61.62%.
2. Network PacketLoss, estimated according to the SDN solution, reduced by 61%
3. By applying the MEC technique in the LTE-A network for the provision of the video
streaming service, the service was delivered with latency reduced by >90 % (0.332s
based to scenario 2 and by 53,996s based to scenario 1), with PacketLoss of almost
0 and with high performance QoE
4. Also, with the increase in the number of End Users, specifically from 20 to 100 UE, and
from 100 to 1000 UE with unchanged packet length of 500 B, unmodified packet delivery
interval 10ms, the latency of delivering video streaming services does not change. In
addition to that, PacketLoss is 0. This is the greatest innovation achieved where the
economic benefits of IPTV Service Providers will be growing up tremendous, while also
increasing in the same time the quality of the delivered IPTV Service.
5. FUTURE WORKS
This research paper has brought about results which are considered as innovations or as
resources to be developed in other studies in the future. Among the main improvements, the
following may be mentioned:
1. Application of MEC technique by giant Video Streaming service providers such as
'Netflix', 'Hulu', 'Amazon Prime', Youtube etc., where the Content could be placed as
close as possible to the users at the borders with the network to deliver the IPTV video
streaming service with low Latency, QoE quality and high performance.
2. Furthermore, a future work could be the construction of optimizing algorithms that will
enable the benefit of IPTV service through SDN based on MEC principles
3. Implementation and production of Virtual Set box (V-Set Box) for the provision of IPTV
and VoD based services to reduce the cost of providing such services and increase the
speed of providing new services to end users.
6. REFERENCES
ETSI, (2016, January 27). Mobile Age Computing. http://www.etsi.org/technologies-clusters/
technologies/mobile-edge-computing.
Hasan, M., Dahshan, H., Abdelwanees, A., &Elmoghazy, A.(2020) SDN Mininet Emulator
Benchmarking and Result Analysis.2nd Novel Intelligent and Leading Emerging Sciences
Conference (NILES), 2020, pp. 355-360, doi: 10.1109/NILES50944.2020.9257913.
Hysenbelliu, E. (2015). A cloud based Architecture for IPTV as a Service. Proceeding ofBallkan
Conference on Informatics: Advanced in ICT, pp.59-64.
Hysenbelliu, E. (2017). GPU Implementation over IPTV Software Defined Networking.Vol. 7 -
Issue 8 August 2017, International Journal of Engineering Research and Applications IJERA ,
ISSN: 2248-9622. www.ijera.com
Hysenbelliu, E. (2017).Toward an Enhanced Quality of IPTV Media Server Architecture over
Software Defined Networking. Vol:11.International Journal of Computer and Information
Engineering, World Academy of Science, Engineering and Technology. ICSED Berlin,
ISNI:0000000091950263. No:5
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ISSN: 2180-1320, https://www.cscjournals.org/journals/IJSE/description.php
Hysenbelliu, E., &Durresi, A (2018). The Evaluation of (CapEx/OpEx) Cost reduction of IPTV
Service delivery using SDN Solution. Volume 20.IOSR Journal of Computer Engineering (IOSR-
JCE) e-ISSN: 2278-0661, p -ISSN: 2278-8727, Issue 5, Ver. I , PP 46-53 www.iosrjournals.org.
Nardini, G.,Virdis, A.,Stea,G.,&Buono, A. (2018).SimuLTE-MEC: Extending SimuLTE for Multi-
Access Edge Computing, EPiC Series in Computing Volume 56, Pages 35-42 Proceedings of the
5th International OMNeT++ Community Sum
Nardini, G., Virdis, A., &Stea,G. (2016). Simulating LTE/LTE-Advanced Networks with
SimuLTE.In: Obaidat M., Ören T., Kacprzyk J., Filipe J. (eds) Simulation and Modeling
Methodologies, Technologies and Applications. Advances in Intelligent Systems and Computing,
vol 402. Springer, Cham. https://doi.org/10.1007/978-3-319-26470-7_5
Nam, H., (2016). Measuring and Improving the Quality of Experience of Adaptive Rate
Vide[Doctoral dissertacion,Columbia University].https://doi.org/10.7916/D82B8Z7V
Sabella, D., Veillant, A., Kurre, A.,Rauschenbach, U.,&Giust, F. (2016). Mobile-Edge Computing
Architecture: The role of MEC in the Internet of Things. IEEE Consumer Electronics
Magazine 5(4):84-91 https://doi.org/:10.1109/MCE.2016.2590118
Sezer, S., Scott-Hayward, S., Chouhan, P. K., Fraser, B., Lake, D., Finnegan, J., Vilijoen, N.,
Miller, M., & Rao, N. (2016). Are We Ready for SDN? Implementation Challenges for Software-
Defined Networks. IEEE Communications Magazine, 51(7), 36-43.
https://doi.org/10.1109/MCOM.2013.6553676
Shahzadi, S., Iqbal, M., &Dagiuklas, T. (2017). Multi-access edge computing: open issues,
challenges and future perspectives. J Cloud Comp 6, 30. https://doi.org/10.1186/s13677-017-
0097-9