QualTest GSM is an Android application that turns your mobile phone into mobile test probe that can make mobile-to-mobile test calls returing AQuA and PVQA call quality metrics including MOS score and audio impairments information.
The document describes a voice control application that allows a user to control their laptop or PC using voice commands. The application uses voice recognition libraries to interpret voice commands and then executes the corresponding functions. It has two components - a PC application that runs in the background and an Android application. The Android app allows remote control of the PC over a network using voice commands recorded on the Android device. The key objectives are to provide full access to the OS without touching the device and enable more interactive control. It also aims to help disabled users control the system using only voice.
IT Essentials (Version 7.0) - ITE Chapter 5 Exam AnswersITExamAnswers.net
油
This document provides answers to exam questions for IT Essentials (ITE v6.0 + v7.0) Chapter 5. It includes 39 multiple choice questions and answers about networking devices, media, protocols, and standards. Key topics covered include switches, routers, wireless access points, Ethernet cabling, TCP/IP model, Wi-Fi standards, and common network ports.
IT Essentials (Version 7.0) - ITE Chapter 6 Exam AnswersITExamAnswers.net
油
The document provides exam answers for IT Essentials (ITE v6.0 + v7.0) Chapter 6. It includes explanations for 24 multiple choice questions covering topics like wireless networking issues, DHCP, static IP addressing, IPv6 addressing and compression, and network troubleshooting steps. Some key issues addressed are weak wireless signals, DHCP server faults, default gateway problems, and IP address conflicts.
This document contains a CCNA exam with multiple choice questions about networking concepts. It covers topics like network layers, protocols, devices, and models. Specifically:
- It asks questions about intermediary devices, network protocols, encapsulation, the OSI model layers, TCP/IP layers, networking devices, network types, and application layer protocols.
- The questions require identifying functions of layers, protocols, devices, network models, and choosing the correct answers related to networking fundamentals.
- Diagrams are included that require identifying elements like network types, devices, and networking processes like encapsulation and multiplexing.
This document provides guidance on measuring LTE KPIs through drive tests and point tests. It outlines the tools, software, and test procedures needed to measure metrics like coverage probability, throughput, latency, and jitter. Drive tests should measure KPIs along roads to test coverage, while point tests evaluate specific metrics like edge throughput and sector throughput at selected locations. The results are used to evaluate whether KPI targets are met.
This document discusses Voice over Internet Protocol (VoIP). It begins with an overview of how traditional telephone networks worked and have evolved to use Internet Protocol. The key protocols and standards used in VoIP like SIP, RTP are explained. The different modes of VoIP like PC-to-PC, PC-to-phone and phone-to-phone are outlined. Advantages of VoIP like cost savings and rich media services are contrasted with disadvantages such as dependence on internet connectivity. Challenges in providing quality of service, addressing delays and security issues with VoIP are also highlighted.
- The document provides an agenda and overview for a technical training on the SETU VTEP device, which is a VoIP-ISDN PRI gateway.
- It describes the device's interfaces, port configuration, hardware architecture, and LED indications for power, reset sequence, SIP trunk status, and T1/E1 port alarms.
- It also covers installation guidelines, applications of the device for VoIP access and PRI gateway functions, and configuration of incoming call routing based on calling number, called number, or DDI number.
The document discusses IMS deployment challenges and solutions at eircom, including:
1. eircom conducted an IMS deployment in 2011 and trial of Rich Communication Services, finding that users liked media sharing and presence capabilities.
2. Post-dial delay was an issue for analog phones connecting through home gateways due to en-bloc dialing; overlap sending was tested as a solution.
3. Ensuring voice quality required benchmarking and simulating congestion to understand degradation; prioritizing voice packets addressed this.
This document provides an overview and instructions for configuring and using a VoIP-FXO-FXS Gateway. It discusses the gateway's interfaces, hardware architecture, LED indicators, installation guidelines, applications, and programming options using a phone or PC. Settings covered include port configuration, incoming/outgoing call management, and advanced options. The gateway allows connecting analog phones to an IP network using SIP protocol and provides voice services over IP for SOHO and small/medium businesses.
The document discusses next generation wireless networks and 4G LTE technology. It provides details on:
1) The requirements of 4G LTE including peak data rates of 100Mbps download and 50Mbps upload, latency under 5ms, mobility support up to 350km/h, and enhanced quality of service.
2) The differences between Frequency Division Duplex (FDD) and Time Division Duplex (TDD) technologies. FDD allows simultaneous upload and download while TDD allows flexible bandwidth allocation.
3) Key components of 4G LTE networks including MIMO, OFDMA, and higher order modulation to achieve higher data rates.
Introduction to SIP(Session Initiation Protocol)William Lee
油
Session Initiation Protocol (SIP) is a signaling protocol for managing multimedia communication sessions over Internet Protocol (IP) networks. SIP can be used to establish two-party or multiparty sessions that include voice, video, chat, gaming, and other forms of media. The document introduces SIP architecture, message format, and common call flows including registration, basic call setup, call modification, call hold, and three-way conferencing.
The Session Initiation Protocol (SIP) is an application-layer signaling protocol used for establishing multimedia sessions over Internet Protocol (IP) networks, such as voice or video calls. SIP can be used to initiate a call, invite participants and manage a call. It defines several methods for call setup, maintenance and termination. Common SIP methods include INVITE for call initiation, ACK to acknowledge call setup, BYE to terminate a call, and REGISTER for registering location. SIP uses SDP for negotiating media capabilities and RTP for transporting media streams.
Machine Learning applications in Voice over IPALTANAI BISHT
油
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
Matrix feature-rich ATAs offer connectivity to VoIP, GSM and POTS networks. An ATA user can plug standard analog telephone devices to the ATA and the analog device(s) will connect transparently to the IP and GSM networks. An ATA thus provides a user with the ease of using a standard telephone instrument, yet make VoIP and GSM calls. The ATAs can also be interfaced to existing PBX system, offering GSM and IP line to be shared among the PBX users.
The Access Point Testing module allows remote testing of wireless access points and the network connectivity seen by wireless clients. It tests connectivity at the wireless, network, transport, and application layers to proactively identify issues. Access point tests can be automated to regularly check connections or used on demand for troubleshooting. Identifying problems early avoids downtime and reduces support costs compared to reactive troubleshooting. The module is part of Motorola's AirDefense Services Platform for comprehensive wireless management.
Ali Naseer has over 8 years of experience as a Support Engineer and Network Support Engineer working on projects involving TCP/IP security and surveillance systems, paging systems, video conferencing solutions, and AV over IP networks. He has a Bachelor's degree in Telecom Engineering from Foundation University in Islamabad and led the installation of an intercommunication system for 54 stations for Pakistan's Punjab Metro System project. His work experience includes positions at Nasco, Motorola, and internships focusing on GSM, WiMax, and RF network troubleshooting.
Guide line tems discovery 3.1 hasp liceseTran Trung
油
This document provides information about licensing for TEMS Discovery software, which uses HASP licensing technology. It discusses local vs. network licenses, license requirements, and inspecting the scope of licenses. It also covers installing local and network HASP HL licenses, using network licenses, restrictions, and updating HASP keys. Network activity for communicating between licensed applications and license servers is described. Supported operating systems for license servers are listed.
This document discusses various techniques for troubleshooting networking issues, including top-down, bottom-up, and divide-and-conquer approaches. It describes using the senses and tools like ipconfig, ping, tracert, netstat, and nslookup to diagnose physical layer problems, connectivity issues, and incorrectly configured devices. Thorough documentation of troubleshooting steps and results is recommended, as is obtaining assistance from outside sources like documentation, forums, or a helpdesk when issues cannot be resolved alone.
This document presents a project to design and implement an IP-based PBX network using open-source software. The system was built on the Elastix server platform with Asterisk PBX as the core. Various hardware components were integrated including an Android smartphone, analog phones, IP phones, and laptops connected over both wired and wireless networks. The system allows for unified communications including voice/video calls, voicemail, IVR, and a database. Compared to traditional PBXs, the IP-PBX provides benefits such as lower costs, easier management, greater scalability and features. Packet loss tests found analog phones had the highest loss while softphones on mobile had the least. Potential applications include use in education, business, healthcare
Remote login allows users to access their work computers from any internet-enabled device. It requires the host computer to be running desktop sharing software and connected to both the internet and a secure network. When a remote login request is made, the desktop sharing software establishes a session between the two computers and exchanges data through a secure channel, allowing the user to access the host computer remotely. Common methods for remote login include SSH for Linux/Unix systems and Telnet, which transforms keystrokes into standard characters that travel over the internet to grant remote terminal access to another system.
This document provides a comprehensive list of port requirements for various SolarWinds products. It includes the port numbers, descriptions of their uses, and in some cases which components or products they are used with. The ports vary between products and some are configurable. Refer to product documentation for specific port information.
IT Essentials (Version 7.0) - ITE Chapter 11 Exam AnswersITExamAnswers.net
油
This document provides the answers to exam questions for IT Essentials (ITE v6.0 + v7.0) Chapter 11. It includes 30 multiple choice questions covering topics like Windows file systems, control panel functions, Internet Explorer configuration, Windows events, RAID levels, boot issues, Active Directory group policy, and Windows file sharing options. The answers are meant to help students prepare for the ITE v6.0 + v7.0 Chapter 11 exam.
Brief introduction into SIP protocol, how it works, common problems to solve. Tech. details about handshake, SIP Trunks and SIP trunking. Market research.
The Session Initiation Protocol (SIP) is the dominant signaling protocol used in VoIP today. It is
responsible for the establishment, control and termination of sessions by exchanging ASCII-text-based
messages between the endpoints. This post goes through the basic components of SIP: messages and
logical entities.
IT Essentials (Version 7.0) - ITE Chapter 12 Exam AnswersITExamAnswers.net
油
This document provides answers to exam questions about mobile device operating systems, security features, and networking. It includes 15 multiple choice questions about topics like:
- Location data sources used by locator apps
- Differences between iOS and Android
- Purposes of passcode locks on mobile devices
- Commands used to backup and store files in Linux
- Methods for removing restrictions from mobile OSs like rooting and jailbreaking
- Safe sources for downloading Android apps like Google Play
Integrate QualTest GSM with desktop or Raspberry Pi. Application receives notification from QualTest test probes about call events, copies recorded calls to desktop, limits time of call, runs pvqa and aqua utilities to estimate voice quality.
Hardware requirements.
Administration manual for Sevana Voice Quality Monitoring solution based on Asterisk PBX. This solutions makes end-to-end voice quality testing and monitoring easy. Various test scenarios for echo or conference birdge testing are already included. AQuA and PVQA impairments analysis together with full VoIP statistics make it suitable for use in any type of network.
- The document provides an agenda and overview for a technical training on the SETU VTEP device, which is a VoIP-ISDN PRI gateway.
- It describes the device's interfaces, port configuration, hardware architecture, and LED indications for power, reset sequence, SIP trunk status, and T1/E1 port alarms.
- It also covers installation guidelines, applications of the device for VoIP access and PRI gateway functions, and configuration of incoming call routing based on calling number, called number, or DDI number.
The document discusses IMS deployment challenges and solutions at eircom, including:
1. eircom conducted an IMS deployment in 2011 and trial of Rich Communication Services, finding that users liked media sharing and presence capabilities.
2. Post-dial delay was an issue for analog phones connecting through home gateways due to en-bloc dialing; overlap sending was tested as a solution.
3. Ensuring voice quality required benchmarking and simulating congestion to understand degradation; prioritizing voice packets addressed this.
This document provides an overview and instructions for configuring and using a VoIP-FXO-FXS Gateway. It discusses the gateway's interfaces, hardware architecture, LED indicators, installation guidelines, applications, and programming options using a phone or PC. Settings covered include port configuration, incoming/outgoing call management, and advanced options. The gateway allows connecting analog phones to an IP network using SIP protocol and provides voice services over IP for SOHO and small/medium businesses.
The document discusses next generation wireless networks and 4G LTE technology. It provides details on:
1) The requirements of 4G LTE including peak data rates of 100Mbps download and 50Mbps upload, latency under 5ms, mobility support up to 350km/h, and enhanced quality of service.
2) The differences between Frequency Division Duplex (FDD) and Time Division Duplex (TDD) technologies. FDD allows simultaneous upload and download while TDD allows flexible bandwidth allocation.
3) Key components of 4G LTE networks including MIMO, OFDMA, and higher order modulation to achieve higher data rates.
Introduction to SIP(Session Initiation Protocol)William Lee
油
Session Initiation Protocol (SIP) is a signaling protocol for managing multimedia communication sessions over Internet Protocol (IP) networks. SIP can be used to establish two-party or multiparty sessions that include voice, video, chat, gaming, and other forms of media. The document introduces SIP architecture, message format, and common call flows including registration, basic call setup, call modification, call hold, and three-way conferencing.
The Session Initiation Protocol (SIP) is an application-layer signaling protocol used for establishing multimedia sessions over Internet Protocol (IP) networks, such as voice or video calls. SIP can be used to initiate a call, invite participants and manage a call. It defines several methods for call setup, maintenance and termination. Common SIP methods include INVITE for call initiation, ACK to acknowledge call setup, BYE to terminate a call, and REGISTER for registering location. SIP uses SDP for negotiating media capabilities and RTP for transporting media streams.
Machine Learning applications in Voice over IPALTANAI BISHT
油
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
Matrix feature-rich ATAs offer connectivity to VoIP, GSM and POTS networks. An ATA user can plug standard analog telephone devices to the ATA and the analog device(s) will connect transparently to the IP and GSM networks. An ATA thus provides a user with the ease of using a standard telephone instrument, yet make VoIP and GSM calls. The ATAs can also be interfaced to existing PBX system, offering GSM and IP line to be shared among the PBX users.
The Access Point Testing module allows remote testing of wireless access points and the network connectivity seen by wireless clients. It tests connectivity at the wireless, network, transport, and application layers to proactively identify issues. Access point tests can be automated to regularly check connections or used on demand for troubleshooting. Identifying problems early avoids downtime and reduces support costs compared to reactive troubleshooting. The module is part of Motorola's AirDefense Services Platform for comprehensive wireless management.
Ali Naseer has over 8 years of experience as a Support Engineer and Network Support Engineer working on projects involving TCP/IP security and surveillance systems, paging systems, video conferencing solutions, and AV over IP networks. He has a Bachelor's degree in Telecom Engineering from Foundation University in Islamabad and led the installation of an intercommunication system for 54 stations for Pakistan's Punjab Metro System project. His work experience includes positions at Nasco, Motorola, and internships focusing on GSM, WiMax, and RF network troubleshooting.
Guide line tems discovery 3.1 hasp liceseTran Trung
油
This document provides information about licensing for TEMS Discovery software, which uses HASP licensing technology. It discusses local vs. network licenses, license requirements, and inspecting the scope of licenses. It also covers installing local and network HASP HL licenses, using network licenses, restrictions, and updating HASP keys. Network activity for communicating between licensed applications and license servers is described. Supported operating systems for license servers are listed.
This document discusses various techniques for troubleshooting networking issues, including top-down, bottom-up, and divide-and-conquer approaches. It describes using the senses and tools like ipconfig, ping, tracert, netstat, and nslookup to diagnose physical layer problems, connectivity issues, and incorrectly configured devices. Thorough documentation of troubleshooting steps and results is recommended, as is obtaining assistance from outside sources like documentation, forums, or a helpdesk when issues cannot be resolved alone.
This document presents a project to design and implement an IP-based PBX network using open-source software. The system was built on the Elastix server platform with Asterisk PBX as the core. Various hardware components were integrated including an Android smartphone, analog phones, IP phones, and laptops connected over both wired and wireless networks. The system allows for unified communications including voice/video calls, voicemail, IVR, and a database. Compared to traditional PBXs, the IP-PBX provides benefits such as lower costs, easier management, greater scalability and features. Packet loss tests found analog phones had the highest loss while softphones on mobile had the least. Potential applications include use in education, business, healthcare
Remote login allows users to access their work computers from any internet-enabled device. It requires the host computer to be running desktop sharing software and connected to both the internet and a secure network. When a remote login request is made, the desktop sharing software establishes a session between the two computers and exchanges data through a secure channel, allowing the user to access the host computer remotely. Common methods for remote login include SSH for Linux/Unix systems and Telnet, which transforms keystrokes into standard characters that travel over the internet to grant remote terminal access to another system.
This document provides a comprehensive list of port requirements for various SolarWinds products. It includes the port numbers, descriptions of their uses, and in some cases which components or products they are used with. The ports vary between products and some are configurable. Refer to product documentation for specific port information.
IT Essentials (Version 7.0) - ITE Chapter 11 Exam AnswersITExamAnswers.net
油
This document provides the answers to exam questions for IT Essentials (ITE v6.0 + v7.0) Chapter 11. It includes 30 multiple choice questions covering topics like Windows file systems, control panel functions, Internet Explorer configuration, Windows events, RAID levels, boot issues, Active Directory group policy, and Windows file sharing options. The answers are meant to help students prepare for the ITE v6.0 + v7.0 Chapter 11 exam.
Brief introduction into SIP protocol, how it works, common problems to solve. Tech. details about handshake, SIP Trunks and SIP trunking. Market research.
The Session Initiation Protocol (SIP) is the dominant signaling protocol used in VoIP today. It is
responsible for the establishment, control and termination of sessions by exchanging ASCII-text-based
messages between the endpoints. This post goes through the basic components of SIP: messages and
logical entities.
IT Essentials (Version 7.0) - ITE Chapter 12 Exam AnswersITExamAnswers.net
油
This document provides answers to exam questions about mobile device operating systems, security features, and networking. It includes 15 multiple choice questions about topics like:
- Location data sources used by locator apps
- Differences between iOS and Android
- Purposes of passcode locks on mobile devices
- Commands used to backup and store files in Linux
- Methods for removing restrictions from mobile OSs like rooting and jailbreaking
- Safe sources for downloading Android apps like Google Play
Integrate QualTest GSM with desktop or Raspberry Pi. Application receives notification from QualTest test probes about call events, copies recorded calls to desktop, limits time of call, runs pvqa and aqua utilities to estimate voice quality.
Hardware requirements.
Administration manual for Sevana Voice Quality Monitoring solution based on Asterisk PBX. This solutions makes end-to-end voice quality testing and monitoring easy. Various test scenarios for echo or conference birdge testing are already included. AQuA and PVQA impairments analysis together with full VoIP statistics make it suitable for use in any type of network.
This document provides information and instructions for performing drivetest, which is a process of collecting network performance data using testing equipment while driving along predetermined routes. It discusses what drivetest is, differences between tuning and optimization, purposes of drivetest, types of drivetest, tools used for drivetest, and best practices for performing drivetest. The document also provides screenshots and exercises for setting up the TEMS Investigation software for drivetest data collection and configuration.
Migration from legacy POTS (plain old telephone system) systems to VoIP technologies has not always proceeded smoothly. The migration must consider many complex issues, including security, reliability, quality of service, and interoperability. Testing is critical to making such migrations smooth and effective. This paper shows issues encountered during migrations along with testing results. It focuses on tests that show the capabilities of a VoIP (voice over Internet protocol) system.
VoLTE Service Monitoring - VoLTE Voice CallJose Gonzalez
油
There is currently no accepted standard for the measurement or monitoring of VoLTE Services, even though we believe that this is vital to assure the quality and reliability of such services - and to establish a framework for reliable comparison across implementations.
To this end Ascom has defined a formal definition and implementation strategy to help the Operations team solve a range of challenges, including issues related to EPC, IMS and the Application Server. We will describe this solution in a number of short articles.
This article describes the architecture of our solution and the VoLTE Voice Call test case.
02 asterisk - the future of telecommunicationsTran Thanh
油
Asterisk is an open-source private branch exchange (PBX) system that can be used to build voice over IP (VoIP) networks and systems. It allows users to reproduce standard PBX features and interact IP-based networks. Asterisk is hardware independent and can run on various operating systems. It provides implementations of basic PBX functionality and integrates with third-party telephony hardware and software.
This document provides release highlights and roadmap information for Mobileum's SITE automation framework. Some key points include:
- The release of new high-level keyword libraries that allow testing of features like VoLTE, audio, SMS, and IP in a more simplified manner.
- The ability to now run automation tests on GRP local units, expanding test capabilities to remote locations.
- Ongoing enhancements to support 5G network slicing testing using different wireless interfaces and emulation of 5G core network interfaces.
- Future roadmap items include enhanced smart device control, automated packet capture analysis, and workflow automation between tests.
The document discusses the design of a cell phone-based home control system that allows users to remotely control and monitor appliances like lights, fans, and thermostats from their phone. It covers the system components, including using text messaging for phone communication, selecting a GM28 cellular module and STK300 microcontroller, and programming the system using C language in Visual Studio. The document also addresses authentication and user interaction methods to securely control devices from the phone.
The document is a resume for an individual seeking a position in a technical organization where they can grow and prove themselves. The individual has over 6 years of experience in the telecom sector as a Testing Engineer, working with products like Nokia IMS and protocols like SIP. They have experience testing various telecom systems and nodes including IMS, LTE, VOIP and more. Their education includes a B.Tech in Electronics and Communication Engineering and they seek to contribute their technical skills and knowledge of telecom protocols.
The document is a resume for an applicant seeking a position in a technical organization. The applicant has over 6 years of experience in the telecom sector working as a Testing Engineer, with expertise in areas like IMS network testing, SIP protocol testing, and system validation. They have comprehensive knowledge of IMS architecture and components like P-CSCF, I-CSCF, S-CSCF, AS, and HSS. The applicant also has experience working with products from Nokia, Ericsson, and other telecom vendors, and has expertise in protocols like SIP, VoIP, and testing tools like Wireshark.
The document provides information on drive testing in GSM networks. Drive testing involves using mobile devices to collect network performance data along predetermined routes. This helps evaluate coverage, availability, capacity, retainability, and call quality from the subscriber perspective. Key aspects discussed include the hardware requirements for drive testing (laptop, data collection software, mobile phones, GPS), different test modes (dedicated call, idle, scan), and important metrics to analyze (Rx level, Rx quality, bit error rate, frame erasure rate, speech quality index). TEMS software is highlighted as a common tool for collecting data and analyzing network performance based on drive test results.
Proving the Security of Low-Level Software Components & TEEsAshley Zupkus
油
Learn how it is possible to prove low-level software component and TEE security, as well as the Goodix driver example demoed in the webinar.
Check out the webinar replay here: https://www.youtube.com/watch?v=nG3DlejBd3k
Visit our website trust-in-soft.com for more information!
The document discusses drive testing using TEMS Investigation software. It provides an overview of the tools needed for drive testing including a laptop, dongle, mobile set, modem, GPS, and more. It outlines the steps to setup the software and ensure all tools are connected and functioning properly. These include attaching the required devices, loading cell files, and selecting the log collection location. The document also describes some key parameters that can be analyzed during drive testing like signal strength, interference, and throughput.
MAF ICIMS Monitoring, Analytics & Reporting for Microsoft Teams and UC - glo...MAF InfoCom
油
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 (SBCs), 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.
The document discusses drive testing using TEMS Investigation software. It provides an overview of the tools needed for drive testing including a laptop, dongle, mobile set, modem, GPS, and more. It outlines the steps to setup the software and ensure all tools are connected and functioning properly. These include attaching the required devices, loading cell files, and selecting the log collection location. The document also describes some key parameters that can be analyzed during drive testing like signal strength, interference, and throughput.
The document describes a specialized software solution called PhaST & T provided by BOSS Informatics for serialization and traceability in drug production. The solution consists of 4 levels - level 1 includes basic hardware, level 2 covers serialization and aggregation software, level 3 includes software on a central server for storage, reports, and synchronization, and level 4 links to external systems like ERP and governmental authorities. The software is designed to provide flexibility, adaptability, low cost, and easy compliance for serialization needs in pharmaceutical manufacturing.
This document summarizes techniques for testing the performance of Asterisk IP PBX servers under heavy call loads. It describes using the Spirent Abacus 5000 tool to generate SIP calls and measure the call setup rate, maximum concurrent calls, latency, packet loss and other metrics. Two Asterisk server configurations were tested: a basic installation and an enterprise installation. The results showed that the enterprise installation could handle more concurrent calls and higher call rates. Proper performance testing and system optimization are important for deploying enterprise-grade Asterisk solutions.
1) Rogue cell towers can manipulate devices by changing network settings and pushing malicious software updates. They have the ability to intercept and redirect all network traffic after compromising a device.
2) A rogue tower can emulate a cell tower and force nearby devices to connect to it. Once connected, it can run scripts to detect device details, push arbitrary code updates, and configure persistent man-in-the-middle attacks by changing APN settings.
3) Field tests showed a rogue tower was able to identify phone models from up to 15 km away and then remotely force software updates that maintained persistent access, even on devices that normally restrict over-the-air updates. However, such attacks are generally not
This document provides instructions for configuring a desktop phone and firewall for use with a 3CX phone system. It discusses supported phone models, provisioning types, configuring BLF and RPS phones, troubleshooting, and using the 3CX web client. It also covers configuring firewall ports, NAT/port settings, SIP trunk providers, inbound and outbound call rules, digital receptionists, and installing 3CX on different operating systems.
This document is a user manual for Sevana AQuA - Audio Quality Analyzer 8.x. It introduces AQuA as a tool for intrusive perceptual voice quality analysis that allows comparing two audio files and testing voice quality loss between a reference and degraded file. The manual describes AQuA's functionality, requirements, testing parameters, scientific background, perceptual modeling, command line usage, and provides examples of comparing audio files and analyzing reasons for voice quality loss.
QualTest mobile test probe for VoIP and mobile call testing and monitoringSevana O端
油
Sevana QualTest is a mobile test probe application that checks
current network conditions and estimates voice quality in mobile and VoIP networks
The platform is designed for end-to-end and single-end call testing, as well as for gathering and analyzing call audio quality metrics. Measure network metrics for VoIP calls and use waveform analysis to correlate audio problems with network conditions.
Real-time monitoring of 5G network. Reliable MOS and other KPIs for messenger-to-messenger and voice calls. Automated mobile-to-mobile testing in 5G networks. Flexible integration and drive testing.
This utility calculates MOS scores for audio streams in .pcap files, optionally decoding the audio to .wav files. It runs on Linux, macOS, and OpenWRT, requires no database, and supports several common codecs. The user provides a .pcap file path and can choose json output or audio saving. The utility then extracts and analyzes RTP streams, calculating MOS scores and statistics and printing the results.
This document discusses using Sevana tools to test call quality between mobile devices using Skype. Specifically, it proposes a demonstration of a Skype-to-Skype call quality test using regular Android Skype apps on mobile phones. The test would record call audio and analyze it using Sevana's AQUA and PVQA tools to generate MOS scores and other quality metrics. It outlines the typical test setup and flow, including host machine, mobile phone, cable adapter, and test management via a backend server.
The document discusses a mobile-to-mobile call quality testing automation solution called Sevana QualTest. It allows for unlimited automated mobile-to-mobile tests without human involvement, saving time compared to manual testing. Key features include active and passive call quality tests using non-rooted Android phones, test result uploads and sharing, and compatibility with all operating systems. Potential use cases include 5G and other mobile network testing, service assurance through quality monitoring, and RAN optimization to improve quality management while reducing call quality testing costs through automation.
Sevana real-time rtp analysis for mobile operatorsSevana O端
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The document describes Sevana's real-time RTP analysis solution for mobile networks. The solution provides call quality monitoring through metrics like MOS and detection of impairments. It allows analysis of call quality for different carriers to evaluate performance, select partners, and identify issues. The solution integrates with existing systems and provides real-time quality monitoring to help optimize networks and prevent churn.
Real time call quality analysis for mobile operatorsSevana O端
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Real waveform analysis for mobile service and solution providers to easily enable:
- Silent call detection
- Audio impairments patterns analysis
- Problem root cause detection
- Fraud detection
- Customized RTP stream interaction
Sevana QualTest is a mobile test probe application which checks current network conditions and estimates voice quality in mobile networks. QualTest is a mobile application for Android
powered devices that can work in both VoIP and cellular networks.
Powerful tool to non-intrusively evaluate
voice quality. Quick and easy setup up of single-ended
voice quality testing. Objective MOS score prediction. Call quality impairments detection.
Sevana PVQA Server is a multipurpose tool for call quality analysis. One can use it as core of their quality analysis and monitoring system to detect and investigate QoE and QoS related issues. PVQA server combines the force of audio waveform analysis using PVQA technology with continuous network monitoring.
Sevana AQuA - Your powerful tool for perceptual voice and audio quality analysis:
Quick and easy setup up of end-to-end voice quality testing
Reliable objective MOS score
Comprehensive waveform analysis
The document describes a real-time RTP call quality monitoring solution using waveform analysis that provides several benefits:
It allows service providers to efficiently manage voice/audio quality in a non-intrusive way based on analyzing the actual media content. This can save on operating expenses by reducing unnecessary payments to partners and prevent revenue loss from customer churn due to poor call quality. The solution provides reliable objective quality metrics and insight into the root causes of quality issues to help justify infrastructure investments. It has high performance and scalability to handle thousands of simultaneous calls through an asynchronous architecture utilizing multiple CPU cores.
This document contains the user manual for Sevana AQuA - Audio Quality Analyzer 7.x. It describes the functionality, requirements, and parameters for using AQuA to compare wav files and test voice quality. Key features include comparing reference files to degraded files, reporting quality scores, analyzing reasons for quality loss, and visualizing signal spectrums. The document also provides details on AQuA's command line parameters and examples of usage.
This document discusses objective quality measurement tools for evaluating voice quality over mobile networks.
It introduces AQuA, an artificial intelligence-based tool that provides more accurate quality scores than PESQ and POLQA by considering factors like codecs, technologies, impairments and languages.
AQuA and PVQA can analyze drive test recordings to generate MOS scores and identify issues like packet loss, hardware problems or network impairments that caused quality degradation.
Tables compare features of PESQ, POLQA and AQuA, showing AQuA can evaluate all audio types and technologies more comprehensively than the other tools.
This document discusses objective quality measurement tools for evaluating voice quality over mobile networks.
It introduces AQuA, an artificial intelligence-based tool that provides more accurate quality scores than PESQ and POLQA by considering factors like codecs, technologies, impairments and languages.
AQuA and PVQA can analyze drive test recordings to generate MOS scores and identify issues like packet loss, hardware problems or network impairments that caused quality degradation.
Tables compare features of PESQ, POLQA and AQuA, showing AQuA evaluates more factors like long calls, stereo audio and custom models, making it best for evaluating VoLTE, VoWiFi and RCS quality during network deployment and operation
AQuA is a tool that uses a model of human hearing to analyze voice signals and make estimates about call quality and the end user's perceived quality. It allows testing of voice quality over different networks and conditions. Sevana provides AQuA with technical support and it can be deployed flexibly as an application, library, or service. AQuA helps monitor quality of services and identify reasons for audio quality issues.
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Qualtest GSM
Document objectives.
This document describes application purpose, requirements, software architecture and user operations for
Qualtest GSM Android application.
Application purpose.
Application provides the following functionality
- measure quality of mobile calls in the field (using rooted Android phones only)
- measure quality of mobile calls with regular non-rooted Android phones and accompanying tools
- upload and share test results
- run both active (with reference audio) and passive tests
- integrate with Qualtest Host and correspondent Backend to automate making and receiving mobile
test calls
Hardware requirements.
CPU ARMv7 or better. x86 support is available upon request.
Memory 1024 MB
Network Any mobile network including 5G.
Software requirements.
Rooted Android OS to run analysis with mobile phone only. Android OS 4.1 and later. Optionally one can
install QualTest Host and Backend. Developer mode MUST be enabled on the phone.
Operation model.
This is a typical Android application with two main use cases:
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- initiate test calls and measure MOS when installed as system application. It is available on rooted
phones only. One can view / share / upload test results right from the phone.
- initiate and accept calls without analysis on mobile phone. Rooted phone is NOT required. In this
mode application notifies its desktop (Raspberry Pi) counterpart (Qualtest Host, further down QH)
about progress of the call. QH handles audio streams via cable adapter and communicates with
backend.
Application installation / uninstallation.
Application is distributed as archive (.zip) with QualtestGSM.apk inside and few Windows and Bash scripts
for installing (and uninstalling) the application on rooted phones.
Installation script will put few Sevana analyzers to /data/local/tmp directory (on mobile device) and will
install QualtestGSM.apk as system application. During the install the application asks to grant superuser
rights on the phone. Phone will be rebooted after installation script finishes working. It is required to
complete installation of Qualtest GSM as system app.
For non-rooted phones usual command adb install QualtestGSM.apk can be used. But in this case
companion application (Qualtest Host) is required.
Application logic.
Main screen represents a list of recorded calls. It includes target number, length of the call, time of the call
end. Recorded call audio files are stored in a dedicated directory (usually named CallRecords). Path to this
directory is shown at the bottom of the screen.
To make analysis just tap any recorded call. The results will be shown on the screen and report files with
.json extension will be written to CallRecords directory. During the analysis Qualtest GSM can request
superuser rights (it is required by license check procedure).
After analysis one can share results via Android share dialog.
Depending on configuration options analysis can run automatically right after the call.
There is Options menu on the main screen. It has the following items:
- Get Host ID. Required to generate system fingerprint file and share it to generate production
license.
- Share. Share call record with analysis results.
- Delete. Removes selected records from the list, deletes corresponding files.
- Upload internal log. Sends internal log to the vendor (Sevana). It is helpful for debugging.
- Settings. Details on the Settings screen one can find further down of the document.
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- Make test call. Initiates mobile call to predefined test number (it can be defined on Settings
screen).
- Upload probe. Uploads call record with analysis results to Qualtest Backend server (QB).
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Settings screen.
List of available parameters:
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Phone instance ID This application instance ID is used for integration with Qualtest
Backend (QB). All reports sent to QB will be bound to this instance
ID. It can be empty, but in this case uploading to QB will not work.
Backend server URL. URL to QB server. It can be empty, but in this case uploading to QB
will not work.
Audio source. Audio source where call is recorded. Usually Voice Downlink is best
choice.
Encoder type. Used encoder type. As Android OS does not let to write plain
uncompressed .wav it is needed to find the best encoder for
further analysis. HE-AAC is quite good; but may not always be the
best option.
Output format. Typically it is kept as Default..
Test number. Target number for tests. It does not restrict QG to this number
only, the application will check ALL calls in the system. However,
this number can be used for quick access from the Options menu.
Limit of recordings. Number of allowed recordings. If new recording exceeds this
number then oldest recording will be removed automatically. Zero
value disables this check.
Run analysis automatically. Analysis will run automatically after each call upon its completion.
It has to be set to allow fully automated tests scheduled via
Qualtest Host.
Audio reference file. Path to reference file. It should be plain uncompressed PCM .wav
file. On some phones selection dialog may have problems and to
avoid that we recommend installing ES File Explorer beforehand.
UNIX timestamps in file
names.
Forces to use UNIX timestamps in call recording file names instead
of full time string. Useful for integration tasks.
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Quality results screen.
Sharing and uploading results is available from the main screen (select desired reports and select Share
from Options menu).
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To see details about PVQA analysis please tap button "Show detectors report"
Detectors report screen:
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