AQuA is a simple but powerful tool to provide perceptual voice quality testing and audio files comparison. This is the easiest way to compare two sound signals and test voice quality between original and degraded files.
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.
AQuA Voice Quality Testing. Competitive Alternative For Pesq (P 862)Sevana Oü
AQuA (Audio Quality Analyzer) is a competitive alternative for existing quality testing models.
AQuA is available for all platforms (32bit and 64bit machines, Windows, Linux, MAC OS)
AQuA has a competitive computational performance
AQuA does not have annual royalty fee
AQuA has the most attractive pricing on the market
AQuA is already used in VoIP PBXs and other systems
Video quality measurements can be performed using subjective, objective, and payload-based methods. Subjective methods involve human assessment while objective methods use measurement devices and are repeatable for testing and monitoring. Payload-based methods assess video quality by comparing the original and distorted video. Standardization bodies have defined various levels of measurement including transport, transaction, and content levels to analyze video quality from different perspectives.
- Voice and video over IP (VoIP/videoconferencing) can work well but requires proper deployment including VLANs, QoS, redundant connections, and bandwidth management.
- Factors like bandwidth, latency, jitter must be considered and can be addressed through QoS tagging and queue management.
- Implementing quality videoconferencing over WAN connections presents challenges of limited bandwidth that can be mitigated using QoS and careful bandwidth allocation.
VoLTE Training Sandton Atma Jaya Jakarta SlideshareAnton Dewantoro
Training on Voice Over LTE (VoLTE) to be conducted at 31 January - 1 February 2015, Atma Jaya University Jakarta.
It covers the signalling flow and CS fallback mechanism!
Audio video ethernet (avb cobra net dante)Jeff Green
AVB fits low-cost, small-form-factor products such as this microphone. The overall trend is that music no longer lives on shelves or in CD racks, but in hard drives in home computers, and increasingly in the cloud. This brings about its own unique problems, not in the encoding system used, or the storage technology, but in distributing the audio from the storage media to the speakers. AVB features are all enabled by a global and port level configuration. Connecting these elements is the AVB-enabled switch (in the graphic above, the Extreme Networks® Summit® X440.) The role of the switch is to provide support for the control protocols: AVB is Ethernet’s next stage of convergence, delivering pitch perfect audio and crystal clear video seamlessly over the network
IP/Ethernet is bringing simplicity and features to audio and video as it has brought to services like VoIP, Storage and many more
High quality, perfectly synchronized A/V until now has been difficult to maintain
Standards work by the IEEE and the AVB standard changes everything, creating interoperability and mass-marketing equipment pricing
Benefits of AVB - Delivers predictable latency and precise synchronization, maximizing the functionality of AV – time synchronization and quality or service
Reduced complexity and Ease of use through interoperability between devices
Streamlines complex network set-up and management, the Infrastructure negotiates and manages the network for optimal prioritized media transport
AV traffic can co-exist with non-AV traffic on same Ethernet infrastructure
Role based control at the XYZ Account - XYZ Account can identify devices and apply policies based on device type all the way down to the port and or the AP. Policies can dynamically change based on the device a user is connecting with and where that user is located. Extreme Networks provides infrastructure to deliver customizable prioritization and scalable capacity via configurable and built-in intelligence, ensuring a comprehensive, superior quality experience. Furthermore, when deployed with Extreme Wireless XYZ Account can configure the network to ensure applications receive the bandwidth they require, while still limiting or preventing high speed streaming of music of video or even games.
There is currently no accepted standard for the measurement or monitoring of RCS 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 1-to-1 Chat test case.
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.
AQuA Voice Quality Testing. Competitive Alternative For Pesq (P 862)Sevana Oü
AQuA (Audio Quality Analyzer) is a competitive alternative for existing quality testing models.
AQuA is available for all platforms (32bit and 64bit machines, Windows, Linux, MAC OS)
AQuA has a competitive computational performance
AQuA does not have annual royalty fee
AQuA has the most attractive pricing on the market
AQuA is already used in VoIP PBXs and other systems
Video quality measurements can be performed using subjective, objective, and payload-based methods. Subjective methods involve human assessment while objective methods use measurement devices and are repeatable for testing and monitoring. Payload-based methods assess video quality by comparing the original and distorted video. Standardization bodies have defined various levels of measurement including transport, transaction, and content levels to analyze video quality from different perspectives.
- Voice and video over IP (VoIP/videoconferencing) can work well but requires proper deployment including VLANs, QoS, redundant connections, and bandwidth management.
- Factors like bandwidth, latency, jitter must be considered and can be addressed through QoS tagging and queue management.
- Implementing quality videoconferencing over WAN connections presents challenges of limited bandwidth that can be mitigated using QoS and careful bandwidth allocation.
VoLTE Training Sandton Atma Jaya Jakarta SlideshareAnton Dewantoro
Training on Voice Over LTE (VoLTE) to be conducted at 31 January - 1 February 2015, Atma Jaya University Jakarta.
It covers the signalling flow and CS fallback mechanism!
Audio video ethernet (avb cobra net dante)Jeff Green
AVB fits low-cost, small-form-factor products such as this microphone. The overall trend is that music no longer lives on shelves or in CD racks, but in hard drives in home computers, and increasingly in the cloud. This brings about its own unique problems, not in the encoding system used, or the storage technology, but in distributing the audio from the storage media to the speakers. AVB features are all enabled by a global and port level configuration. Connecting these elements is the AVB-enabled switch (in the graphic above, the Extreme Networks® Summit® X440.) The role of the switch is to provide support for the control protocols: AVB is Ethernet’s next stage of convergence, delivering pitch perfect audio and crystal clear video seamlessly over the network
IP/Ethernet is bringing simplicity and features to audio and video as it has brought to services like VoIP, Storage and many more
High quality, perfectly synchronized A/V until now has been difficult to maintain
Standards work by the IEEE and the AVB standard changes everything, creating interoperability and mass-marketing equipment pricing
Benefits of AVB - Delivers predictable latency and precise synchronization, maximizing the functionality of AV – time synchronization and quality or service
Reduced complexity and Ease of use through interoperability between devices
Streamlines complex network set-up and management, the Infrastructure negotiates and manages the network for optimal prioritized media transport
AV traffic can co-exist with non-AV traffic on same Ethernet infrastructure
Role based control at the XYZ Account - XYZ Account can identify devices and apply policies based on device type all the way down to the port and or the AP. Policies can dynamically change based on the device a user is connecting with and where that user is located. Extreme Networks provides infrastructure to deliver customizable prioritization and scalable capacity via configurable and built-in intelligence, ensuring a comprehensive, superior quality experience. Furthermore, when deployed with Extreme Wireless XYZ Account can configure the network to ensure applications receive the bandwidth they require, while still limiting or preventing high speed streaming of music of video or even games.
There is currently no accepted standard for the measurement or monitoring of RCS 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 1-to-1 Chat test case.
This document discusses assessing network readiness for audiovisual systems. It covers service level agreements (SLAs), which are contracts that define service targets for bandwidth, latency, and packet loss. Effective SLAs have components like needs analysis, roles and responsibilities, and methods for measurement and review. The document also discusses setting service targets for bandwidth usage, latency, and packet loss. Finally, it addresses network ports, protocols, firewalls, and documenting port and protocol requirements for devices.
Syed Idrish has over 2 years of experience in network support roles. He currently works at Samsung Electronics troubleshooting backhaul issues like low throughput and VoLTE call quality. Previously he worked at Tikona Digital Network and Vedang Cellular Service in network support and project engineering roles. He has skills in Linux, routing protocols, switching, and troubleshooting networking issues. He is pursuing a CCNP in routing and switching and holds a BE in Electronics and Communications.
- Adaptive Multi-Rate (AMR) speech coding allows variable bit rates depending on channel conditions between 4.75 kbit/s and 12.2 kbit/s over full-rate channels and between 4.75 kbit/s and 7.95 kbit/s over half-rate channels. It uses source coding, channel coding, and rate adaptation based on channel estimation to optimize quality and efficiency.
- AMR utilizes algebraic code excited linear prediction speech coding, unequal error protection, recursive systematic convolutional channel coding, and discontinuous transmission with voice activity detection and comfort noise generation. This allows it to save bandwidth during silence and adapt to changing channel conditions.
This document summarizes a tutorial on video over 802.11 networks. It discusses motivations for using 802.11 for video, outlines various use cases and their requirements. It then covers challenges of transmitting video over wireless like interference, limited channels and non-deterministic medium access. Current 802.11 mechanisms for video are outlined along with their limitations. Possible areas for further work are identified like content-aware techniques and inter-layer communication. Related activities outside 802.11 are also briefly mentioned.
Lte drivetest guideline with genex probeRay KHASTUR
1. Drive testing involves collecting radio frequency data through a test device to evaluate network coverage and quality of service parameters.
2. Major LTE key performance indicators measured include accessibility, retainability, mobility, and integrity. Key parameters measured are RSRP, SINR, and throughput.
3. To perform an effective drive test, the tester must understand parameter definitions, avoid duplicate routes, analyze events like call drops, and strategically test throughput in stable conditions.
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.
Currently no accepted standard for measurement, monitoring VoLTE Services.
Ascom has defined a formal definition, implementation strategy to help Operations.
Covers issues related to EPC, IMS, Application Service.
VoIP Monitoring and Analysis - Still Top of Mind in Network Performance Monit...Savvius, Inc
With over 10 years of deployment history, VoIP is the primary voice solution for just about every company in existence - large, medium, or small. But even with all that history, recent research from TRAC shows that VoIP is still the number one IT initiative impacting network performance. And with the growth of 802.11 and Wi-Fi enabled smart phones, the use of voice over Wi-Fi (VoFi) promises to increase the volume of VoIP traffic even more. Analyzing VoIP traffic alone is not enough. VoIP analysis must be part of your overall network performance analysis. After all, VoIP is just another data type on your network, and according to TRAC, it is impacting your network performance, so you must monitor and analyze the network as a whole, including voice and video over IP. Join us to see how easy it is to capture and analyze voice, video, and data traffic simultaneously, allowing you to pinpoint the impact of each data type on your overall network performance.
Audio Essentials for Broadcast and MultiscreenEllis Reid
This wall chart highlights the key terms and standards required for the delivery of premium audio across broadcast and multiscreen workflows. It is designed as a quick reference for people who are responsible for delivering rich media experiences acress broadcast and over the top networks
Spectrum management best practices in a Gigabit wireless worldCisco Canada
With the introduction of 802.11ac the news is full of the potential for Gigabit networking. Very few of us will have the luxury of running a network that strictly supports 802.11ac and that means a mixed environment for most of us. Get the facts on what 802.11ac means to you, how to evaluate using 20, 40, 80 or 160 Mhz OBSS/Channels. How does RRM's DCA handle a mixed environment and what performance considerations do you need to consider to make decisions that make the best of the spectrum you have today and in the future. What is in the future for our spectrum? To learn more please visit our website here: http://www.cisco.com/ca/
Overview of VoIP (Voice over IP) and FoIP (Fax over IP) technologies like Session Initiation Protocol and H.323.
Even though voice over IP (VoIP) was hailed as a technological innovation, the idea to transport real-time traffic over TCP/IP networks was not new back in the 1990s when VoIP started being deployed in networks. Chapter 2.5 of the venerable RFC793 (TCP) shows both data oriented application traffic as well as voice being transported over IP based networks.
Nevertheless, VoIP puts high demands on signal and protocol processing capabilities so it became possible at reasonable costs only in the 1990s.
VoIP can be roughly split into two main functions. Signaling protocols like SIP (Session Initiation Protocol), H.323 and MGCP/H.248 are used to establish a conference session and the data path for transporting real-time voice data packets. SIP has largely supplanted H.323 in recent years to its simpler structure and packet sequences. MGCP and H.248 are mostly used in carrier backbone networks.
Protocols like RTP (Real Time Protocol) transport voice packets and provide the necessary information for receivers to equalize packet flow variations to provide a smooth playback of the original voice signal.
Voice codecs are one of the core functions of the data path. Voice compression reduces the bandwidth required to transport voice over an IP based network. Compression may be less of a concern in local area networks with gigabit speeds, on slower links like 3G (UMTS, LTE) it still makes a lot of sense.
The algorithms used in different codecs make use of various characteristics of the characteristics of human speech recognition. Redundant information is removed from the signals thus slightly reducing the quality, but greatly reducing the required bandwidth.
In VoIP networks, the echo problem is typically compounded by the increased delay incurred by packetization of voice signals. To counteract the echo problem, VoIP gear (hard phones, soft phones, gateways) include echo cancelers to remove echo signals from the transmit signal.
To transport facsimile over an IP based network, even more technology is needed. Facsimile protocols are very susceptible to delay and delay variation and thus need more compensation algorithms. Protocols like T.38 terminate facsimile protocols like T.30 (analog facsimile) and transport the fax images as digitized pictures over IP based networks.
Practical Fundamentals of Voice over IP (VoIP) for Engineers and TechniciansLiving Online
This manual provides solid practical advice on application, implementation and, most importantly, troubleshooting Voice Over IP (VOIP) systems.
MORE INFORMATION: http://www.idc-online.com/content/practical-fundamentals-voice-over-ip-voip-21?id=151
The document compares four video conferencing software programs: Microsoft NetMeeting, Cu-SeeMe, Ivisit, and HoneyCom. It analyzes each program based on audio and video quality, time delay, bandwidth, and packet loss for connections between Albuquerque, New Mexico and other locations. Overall, Cu-SeeMe performed the best while HoneyCom performed the worst, especially for connections over long distances.
VoLTE Service Monitoring - VoLTE Voice CallJose Gonzalez
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.
Practical Fundamentals of Voice over IP (VoIP) for Engineers and TechniciansLiving Online
In the past five years, technologies have converged to such an extent that one can transmit voice, fax and video over the same internet protocol network that one uses for data. This workshop examines Voice over IP (VoIP) technologies and provides you with the skills to competently implement a VoIP network for your organisation. Numerous case studies and exercises throughout the course ensure that you get a good grasp on the technologies used. Solid practical advice is given on application, implementation and most importantly troubleshooting these systems.
MORE INFORMATION: http://www.idc-online.com/content/practical-fundamentals-voice-over-ip-voip-engineers-and-technicians-3
AMR is a new speech coding standard that allows for adaptive multi-rate coding. It uses multiple coding rates from 4.75 kbps to 12.2 kbps depending on radio frequency conditions, using lower rates with more error protection in poor conditions. This provides improved speech quality overall compared to existing standards. AMR also enables potential cell extension since lower rates provide more robust links. It allows for half-rate modes for increased network capacity. Motorola infrastructure requires new hardware to support AMR and half-rate modes.
This document summarizes a student project on Voice over IP (VoIP) quality of service. It discusses how VoIP works by converting analog speech to digital packets sent over the Internet. It then covers current Internet limitations for real-time applications like VoIP. It evaluates scheduling algorithms like FIFO, priority queueing, and weighted fair queueing. The document outlines simulating these algorithms in OPNET and analyzing results. Based on this, it proposes a new algorithm using priority queueing for real-time traffic and weighted fair queueing with dynamic weights for other traffic. Simulation results show the proposed algorithm meeting quality of service requirements for different traffic classes.
This document discusses various key performance indicators (KPIs) for Voice over LTE (VoLTE) networks. It describes KPIs for VoLTE control plane performance like registration success rate, call setup success rate, and call setup time. It also covers user plane KPIs such as mute rate, mean opinion score, RTP packet loss rate, and one way call rate. Additionally, it lists KPIs for packet core network elements like attach success rate, paging success rate, and IP pool utilization. The document provides details on calculating each KPI and healthy range benchmarks.
This document describes the measurement configurations and procedures for DAB receiver testing using National Instruments PXI Vector Signal Generators and MaxEye DAB/DABPlus/T-DMB Signal Generation software.
Voice over IP (VoIP) Speech Quality Measurement with Open-Source Software Com...Sebastian Schumann
This paper proposes an alternative to expensive means for VoIP speech quality measurement. While current applications and measurement devices on the market are very expensive, the authors propose a solution based on open-source components that allows the determination of the Mean Opinion Score (MOS) value according the Perceptual Evaluation of Speech Quality (PESQ) test methodology. Presented at Elmar 2010 in Zadar, Croatia.
The document discusses testing telephony applications. It recommends unit testing application components like models in isolation. For integration testing, it suggests that automated testing is needed due to the complex scenarios and branching flows, but commercial options are expensive. It then introduces Cucumber-VoIP, an open source framework for automating integration tests of the full stack including voice interactions and media.
This document discusses assessing network readiness for audiovisual systems. It covers service level agreements (SLAs), which are contracts that define service targets for bandwidth, latency, and packet loss. Effective SLAs have components like needs analysis, roles and responsibilities, and methods for measurement and review. The document also discusses setting service targets for bandwidth usage, latency, and packet loss. Finally, it addresses network ports, protocols, firewalls, and documenting port and protocol requirements for devices.
Syed Idrish has over 2 years of experience in network support roles. He currently works at Samsung Electronics troubleshooting backhaul issues like low throughput and VoLTE call quality. Previously he worked at Tikona Digital Network and Vedang Cellular Service in network support and project engineering roles. He has skills in Linux, routing protocols, switching, and troubleshooting networking issues. He is pursuing a CCNP in routing and switching and holds a BE in Electronics and Communications.
- Adaptive Multi-Rate (AMR) speech coding allows variable bit rates depending on channel conditions between 4.75 kbit/s and 12.2 kbit/s over full-rate channels and between 4.75 kbit/s and 7.95 kbit/s over half-rate channels. It uses source coding, channel coding, and rate adaptation based on channel estimation to optimize quality and efficiency.
- AMR utilizes algebraic code excited linear prediction speech coding, unequal error protection, recursive systematic convolutional channel coding, and discontinuous transmission with voice activity detection and comfort noise generation. This allows it to save bandwidth during silence and adapt to changing channel conditions.
This document summarizes a tutorial on video over 802.11 networks. It discusses motivations for using 802.11 for video, outlines various use cases and their requirements. It then covers challenges of transmitting video over wireless like interference, limited channels and non-deterministic medium access. Current 802.11 mechanisms for video are outlined along with their limitations. Possible areas for further work are identified like content-aware techniques and inter-layer communication. Related activities outside 802.11 are also briefly mentioned.
Lte drivetest guideline with genex probeRay KHASTUR
1. Drive testing involves collecting radio frequency data through a test device to evaluate network coverage and quality of service parameters.
2. Major LTE key performance indicators measured include accessibility, retainability, mobility, and integrity. Key parameters measured are RSRP, SINR, and throughput.
3. To perform an effective drive test, the tester must understand parameter definitions, avoid duplicate routes, analyze events like call drops, and strategically test throughput in stable conditions.
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.
Currently no accepted standard for measurement, monitoring VoLTE Services.
Ascom has defined a formal definition, implementation strategy to help Operations.
Covers issues related to EPC, IMS, Application Service.
VoIP Monitoring and Analysis - Still Top of Mind in Network Performance Monit...Savvius, Inc
With over 10 years of deployment history, VoIP is the primary voice solution for just about every company in existence - large, medium, or small. But even with all that history, recent research from TRAC shows that VoIP is still the number one IT initiative impacting network performance. And with the growth of 802.11 and Wi-Fi enabled smart phones, the use of voice over Wi-Fi (VoFi) promises to increase the volume of VoIP traffic even more. Analyzing VoIP traffic alone is not enough. VoIP analysis must be part of your overall network performance analysis. After all, VoIP is just another data type on your network, and according to TRAC, it is impacting your network performance, so you must monitor and analyze the network as a whole, including voice and video over IP. Join us to see how easy it is to capture and analyze voice, video, and data traffic simultaneously, allowing you to pinpoint the impact of each data type on your overall network performance.
Audio Essentials for Broadcast and MultiscreenEllis Reid
This wall chart highlights the key terms and standards required for the delivery of premium audio across broadcast and multiscreen workflows. It is designed as a quick reference for people who are responsible for delivering rich media experiences acress broadcast and over the top networks
Spectrum management best practices in a Gigabit wireless worldCisco Canada
With the introduction of 802.11ac the news is full of the potential for Gigabit networking. Very few of us will have the luxury of running a network that strictly supports 802.11ac and that means a mixed environment for most of us. Get the facts on what 802.11ac means to you, how to evaluate using 20, 40, 80 or 160 Mhz OBSS/Channels. How does RRM's DCA handle a mixed environment and what performance considerations do you need to consider to make decisions that make the best of the spectrum you have today and in the future. What is in the future for our spectrum? To learn more please visit our website here: http://www.cisco.com/ca/
Overview of VoIP (Voice over IP) and FoIP (Fax over IP) technologies like Session Initiation Protocol and H.323.
Even though voice over IP (VoIP) was hailed as a technological innovation, the idea to transport real-time traffic over TCP/IP networks was not new back in the 1990s when VoIP started being deployed in networks. Chapter 2.5 of the venerable RFC793 (TCP) shows both data oriented application traffic as well as voice being transported over IP based networks.
Nevertheless, VoIP puts high demands on signal and protocol processing capabilities so it became possible at reasonable costs only in the 1990s.
VoIP can be roughly split into two main functions. Signaling protocols like SIP (Session Initiation Protocol), H.323 and MGCP/H.248 are used to establish a conference session and the data path for transporting real-time voice data packets. SIP has largely supplanted H.323 in recent years to its simpler structure and packet sequences. MGCP and H.248 are mostly used in carrier backbone networks.
Protocols like RTP (Real Time Protocol) transport voice packets and provide the necessary information for receivers to equalize packet flow variations to provide a smooth playback of the original voice signal.
Voice codecs are one of the core functions of the data path. Voice compression reduces the bandwidth required to transport voice over an IP based network. Compression may be less of a concern in local area networks with gigabit speeds, on slower links like 3G (UMTS, LTE) it still makes a lot of sense.
The algorithms used in different codecs make use of various characteristics of the characteristics of human speech recognition. Redundant information is removed from the signals thus slightly reducing the quality, but greatly reducing the required bandwidth.
In VoIP networks, the echo problem is typically compounded by the increased delay incurred by packetization of voice signals. To counteract the echo problem, VoIP gear (hard phones, soft phones, gateways) include echo cancelers to remove echo signals from the transmit signal.
To transport facsimile over an IP based network, even more technology is needed. Facsimile protocols are very susceptible to delay and delay variation and thus need more compensation algorithms. Protocols like T.38 terminate facsimile protocols like T.30 (analog facsimile) and transport the fax images as digitized pictures over IP based networks.
Practical Fundamentals of Voice over IP (VoIP) for Engineers and TechniciansLiving Online
This manual provides solid practical advice on application, implementation and, most importantly, troubleshooting Voice Over IP (VOIP) systems.
MORE INFORMATION: http://www.idc-online.com/content/practical-fundamentals-voice-over-ip-voip-21?id=151
The document compares four video conferencing software programs: Microsoft NetMeeting, Cu-SeeMe, Ivisit, and HoneyCom. It analyzes each program based on audio and video quality, time delay, bandwidth, and packet loss for connections between Albuquerque, New Mexico and other locations. Overall, Cu-SeeMe performed the best while HoneyCom performed the worst, especially for connections over long distances.
VoLTE Service Monitoring - VoLTE Voice CallJose Gonzalez
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.
Practical Fundamentals of Voice over IP (VoIP) for Engineers and TechniciansLiving Online
In the past five years, technologies have converged to such an extent that one can transmit voice, fax and video over the same internet protocol network that one uses for data. This workshop examines Voice over IP (VoIP) technologies and provides you with the skills to competently implement a VoIP network for your organisation. Numerous case studies and exercises throughout the course ensure that you get a good grasp on the technologies used. Solid practical advice is given on application, implementation and most importantly troubleshooting these systems.
MORE INFORMATION: http://www.idc-online.com/content/practical-fundamentals-voice-over-ip-voip-engineers-and-technicians-3
AMR is a new speech coding standard that allows for adaptive multi-rate coding. It uses multiple coding rates from 4.75 kbps to 12.2 kbps depending on radio frequency conditions, using lower rates with more error protection in poor conditions. This provides improved speech quality overall compared to existing standards. AMR also enables potential cell extension since lower rates provide more robust links. It allows for half-rate modes for increased network capacity. Motorola infrastructure requires new hardware to support AMR and half-rate modes.
This document summarizes a student project on Voice over IP (VoIP) quality of service. It discusses how VoIP works by converting analog speech to digital packets sent over the Internet. It then covers current Internet limitations for real-time applications like VoIP. It evaluates scheduling algorithms like FIFO, priority queueing, and weighted fair queueing. The document outlines simulating these algorithms in OPNET and analyzing results. Based on this, it proposes a new algorithm using priority queueing for real-time traffic and weighted fair queueing with dynamic weights for other traffic. Simulation results show the proposed algorithm meeting quality of service requirements for different traffic classes.
This document discusses various key performance indicators (KPIs) for Voice over LTE (VoLTE) networks. It describes KPIs for VoLTE control plane performance like registration success rate, call setup success rate, and call setup time. It also covers user plane KPIs such as mute rate, mean opinion score, RTP packet loss rate, and one way call rate. Additionally, it lists KPIs for packet core network elements like attach success rate, paging success rate, and IP pool utilization. The document provides details on calculating each KPI and healthy range benchmarks.
This document describes the measurement configurations and procedures for DAB receiver testing using National Instruments PXI Vector Signal Generators and MaxEye DAB/DABPlus/T-DMB Signal Generation software.
Voice over IP (VoIP) Speech Quality Measurement with Open-Source Software Com...Sebastian Schumann
This paper proposes an alternative to expensive means for VoIP speech quality measurement. While current applications and measurement devices on the market are very expensive, the authors propose a solution based on open-source components that allows the determination of the Mean Opinion Score (MOS) value according the Perceptual Evaluation of Speech Quality (PESQ) test methodology. Presented at Elmar 2010 in Zadar, Croatia.
The document discusses testing telephony applications. It recommends unit testing application components like models in isolation. For integration testing, it suggests that automated testing is needed due to the complex scenarios and branching flows, but commercial options are expensive. It then introduces Cucumber-VoIP, an open source framework for automating integration tests of the full stack including voice interactions and media.
Pallavi Verma completed an internship as a Business Analyst Intern at Fiserv, a financial services technology company. During her internship, she worked on two projects - migrating an interactive voice response (IVR) system to the cloud, and developing a cost model to consolidate expenses related to the IVR system. She also assisted with project management tasks like tracking deadlines and coordinating with infrastructure teams.
This document discusses best practices for building world-class voice applications (IVRs). It begins by explaining why people dislike IVRs and argues that people dislike bad automation, not self-service itself. It then outlines three key challenges with IVRs: they impose themselves on users, have a constrained voice interface, and must simulate human behavior. The document advocates for a "Caller First" philosophy of rapid development, leveraging caller and application data, and intelligent behavior. Examples of intelligent behavior include avoiding main menus, prioritizing options, and bypassing automation for unhappy callers. The presentation concludes by emphasizing the importance of data and putting the caller first.
The document compares three testing frameworks: data-driven, keyword-driven, and hybrid. In a data-driven framework, test scripts read data from an Excel file and write results to another Excel file. A keyword-driven framework uses a test steps Excel file to read keywords and write test cases. A hybrid framework combines elements of both data-driven and keyword-driven frameworks by allowing test scripts to read both keywords and data.
WSO2 Test Automation Framework : Approach and AdoptionWSO2
The document discusses WSO2's test automation framework. It provides an overview of the framework's requirements, history, approach, architecture, and structure. The framework is designed to make automation easy, organized, relevant and optimized. It supports integration, platform, and cloud-based deployment testing across multiple WSO2 products. Key aspects include flexible execution modes, a context provider, server management, the automation API, utilities, and reporting. Challenges and future areas of focus are also presented, along with demonstrations of sample tests.
Moodle is structured with a site at the top level containing categories to organize courses. Courses are then comprised of topics, resources, activities, and blocks. All parts of Moodle including the site, categories, courses, and their components are considered contexts where user roles can be assigned. The document outlines this structure and emphasizes that users have roles within contexts rather than being assigned globally in Moodle.
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Automatic Sound Signals Quality Estimation Business BenefitsSevana Oü
This presentation describes benefits for business in choosing Sevana voice quality analysis software. It\'s hard to believe that state-of-the-art demanding solution does exist, but one who does it first wins and "the winner takes it all!"
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Sevana provides voice quality analysis software to help companies ensure high quality delivery to customers. The software can analyze voice quality over fixed, mobile and IP networks even under adverse conditions. It allows developers and providers to test voice processing technologies without worrying about means for quality testing. The software is optimized for various platforms and provides flexible integration and testing of voice codecs and quality in VoIP environments.
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The document provides instructions for setting up an AudioInspector demo, including downloading client software from their website without installation, logging in with a provided access code, allowing remote access and control of the software over chat or VoIP, and adjusting audio settings if needed. It then summarizes the AudioInspector product family which performs automated audio file inspection, quality assessment, and metadata processing, and provides an example archive workflow and list of its key features.
- Digium created Asterisk, the open source telephony platform, and is the leading provider of open source business phone systems through products like Switchvox.
- Switchvox is an all-in-one business phone system that provides features like call queues, conferencing, call recording, and more through an easy-to-use web interface.
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Netwave Telecom is a Voice over Internet Protocol telecommunications company that offers phone systems and services for small and medium-sized businesses. It provides an affordable VoIP solution with features such as auto attendants, conferencing, voicemail, and video conferencing. While VoIP has advantages over traditional phone systems in terms of features and costs, it also has some disadvantages such as loss of service during power outages and potential voice quality issues that are reduced with sufficient internet bandwidth.
This document summarizes hosted VoIP services provided by Reignmaker Communications. It defines VoIP and hosted VoIP, explaining that hosted VoIP provides phone system features over the internet rather than traditional phone lines. It outlines key advantages like lower costs, scalability, and built-in disaster recovery. Finally, it introduces Reignmaker's service offerings and positioning as an end-to-end provider with nationwide coverage and exceptional support.
Global headquarters in Bedford, MA USA with regional headquarters in Europe (UK) and Asia (Japan). Founded in 1992 originally as Hammer Technologies, Empirix leads the market in service quality assurance solutions for new IP communications. Empirix helps customers ensure quality of experience, avoid outages, minimize downtime impact, and gain returns through improved satisfaction and reduced costs.
Telepresence success is based on an almost flawless presentation of HD video and Audio. Clearly the video and audio quality are clearly the most relevant metrics. But in relation to understanding the full picture of quality of experience more test parameters need to be included in the test strategy. Evidently Telepresence is not the only service to run on a network, therefore its important to test with varying applications and load 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.
Hosted PBX- Should You Be a Provider or a Reseller?NetSapiens
The growing demands for Hosted PBX services in the SMB and Enterprise markets have opened up profitable opportunities for service providers. It is easy to decide to capture these opportunities but how should you build your business model? In this webinar, we will discuss the differences in the two most popular strategies for entering into the Hosted PBX space and answer the question that many growing ITSPs have on their mind: "Should I be a provider or a reseller?"
The document discusses ShoreTel's IP telephony system and its advantages over traditional digital PBX systems. It highlights ShoreTel's ease of use, high customer satisfaction ratings, low total cost of ownership, reliability, and ability to provide a single communications platform across multiple business sites. The system allows businesses to have standardized phone features and disaster recovery capabilities while minimizing server usage for lower costs compared to server-centric competitors. ShoreTel has over 5,000 customers across various industries that have benefited from features such as call conferencing, mobility, and increased productivity.
RealSpeaker is an audio-visual speech recognition system that uses video information in addition to audio to improve accuracy rates by 20-30% over traditional speech recognition. It offers more security through audio-video verification and is cheaper than competing products like Nuance. RealSpeaker allows users to enter text using their voice without a keyboard and supports over 13 languages.
The document discusses using traffic emulation to test the interoperability and performance of telepresence systems. It describes emulating real end users and devices to conduct tests that evaluate factors like video and audio quality, call connectivity, and system functionality under different conditions. Traffic emulation allows for testing scenarios that incorporate real user behavior and diverse network environments.
The document introduces the UniPhyer network switch, which aims to simplify voice and data convergence. It provides a dedicated voice path over existing telephone wiring to guarantee quality of service for voice, separate from the data network. This allows customers to benefit from IP telephony without compromising voice quality or disrupting their existing infrastructure. The UniPhyer can reportedly reduce costs, risks, and management complexity compared to traditional converged network approaches.
Voice and Video over IP Communications: Assessing and Improving User ExperienceRADVISION Ltd.
While video deployment over IP is experiencing a significant boom, overall user experience does not always live up to expectations. RADVISION experts will discuss unique, no-reference, video measurement and analysis algorithms that improve user experience for voice and video over IP communications.
By attending this webinar, you will learn:
* Factors that can affects Video Quality
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* RADVISION solutions for Video Quality
Who should attend: professionals in visual communication field
* IT managers
* System architects
* Product Managers
* CTOs
* VP R&D
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How to build a personalized IVR with DTMF and SpeechEnablex1
Interactive Voice Response (IVR) is a powerful technology which enables businesses to provide exceptional customer service through automated assistance, routing, and information. IVR systems can process Speech and touch-tone inputs from callers, making it easy to streamline processes, reduce costs, and improve customer satisfaction. Know More: https://www.enablex.io/insights/how-to-build-a-personalized-ivr-with-dtmf-and-speech/
Transcend is a value-added partner of ShoreTel that has been delivering business solutions since 1984. It has a methodical sales and implementation process to ensure high quality and has over 95% customer satisfaction. Transcend provides a range of IP telephony, network, security, and video solutions from vendors such as ShoreTel, Cisco, and Avaya to meet customer needs.
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Own and Operate your very own VoIP Platform. Let Comm-Core show you how to earn profits of 80 to 90% each month from your VoIP Customers and still save them 25% or more!
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Overview
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What to expect
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Folding is a recent technique for building efficient recursive SNARKs. Several elegant folding protocols have been proposed, such as Nova, Supernova, Hypernova, Protostar, and others. However, all of them rely on an additively homomorphic commitment scheme based on discrete log, and are therefore not post-quantum secure. In this work we present LatticeFold, the first lattice-based folding protocol based on the Module SIS problem. This folding protocol naturally leads to an efficient recursive lattice-based SNARK and an efficient PCD scheme. LatticeFold supports folding low-degree relations, such as R1CS, as well as high-degree relations, such as CCS. The key challenge is to construct a secure folding protocol that works with the Ajtai commitment scheme. The difficulty, is ensuring that extracted witnesses are low norm through many rounds of folding. We present a novel technique using the sumcheck protocol to ensure that extracted witnesses are always low norm no matter how many rounds of folding are used. Our evaluation of the final proof system suggests that it is as performant as Hypernova, while providing post-quantum security.
Paper Link: https://eprint.iacr.org/2024/257
zkStudyClub - LatticeFold: A Lattice-based Folding Scheme and its Application...
Automated Voice And Audio Quality Test Measurement
1. Making your voice quality testing easy Sevana Oy Endre Domiczi Mobile: +358 9 23164165 E-mail: ceo@sevana.fi
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3. Company Presentation Sevana Oy (Ltd) is a privately owned Helsinki based Finnish company established in 2003 as a software development company with the emphasis to emerging software technologies in telecom and industrial solutions. Sevana Oy Agrikolankatu 11 00530 Helsinki email: info@sevana.fi Finland tel. : +358 9 23164165
4. Company management and shareholders Sevana Oy (Ltd.) is currently owned by its founders and employees. The company has three different international software research and development teams organized by product lines. Sevana Oy (Ltd.) is lead by Mr. Endre Domiczi company CEO who is one of the main shareholders and co-founders. He is currently responsible for company operations and managing activities in Finland and European markets.
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18. Business Solutions(4): VQT in Mobile Networks Mobile device with a stored voice/audio of original quality initiating a call to “answering machine” Mobile network of any standard providing carrier to the “answering machine” and being reason for audio degradation “ Answering machine” when receiving a call plays back the same original audio to the mobile device Test network quality for voice transmission from a Windows Mobile enabled device in seconds! Mobile Network Originating a phone call Answering to phone call Playing recorded audio Receiving degraded sound
19. Business Solutions(5): VQ Monitoring Computer generating and storing “speech model” – a set of sounds typical for human speech – delivering to the reception side through carrier channel/trunk Network carrier: PSTN/GSM/3G/CDMA/IP Computer receiving degraded “speech model” and calculating voice quality value in % plus reasons for voice quality loss Test channels/trunks for voice quality continuously. No other equipment required than what you already have! Carrier Network Speech Model Signal Degraded Speech Model Signal
20. Business Solutions(6): VoIP: QoE Feedback Avoid asking VoIP users for voice quality feedback, automate receiving perceptual quality of experience values – make your system more customer friendly! On one end of VoIP conversation outgoing (original) and incoming (degraded) are recorded On the other end of VoIP conversation outgoing (original) and incoming (degraded) are recorded as well VoIP Network VQ Server tests original1 to degraded2 and degraded1 to original1 providing 2 values for voice quality of the conversation
21. Business Solutions(7): VoIP: QoS Monitor your VoIP provider by doing sudden tests and be aware that your provider maintains appropriate network and voice quality (especially if you signed SLA) Initiating a call would requite just the same client device/application and 1 license of Sevana Voice Quality Software Test voice quality in your VoIP environment and get back to your VoIP provider with questions on SLA and reasons for voice quality loss – you will be prepared with estimations and actual reasons what degrades voice quality in your network. VoIP network conditions as well as voice quality conditions may vary, but when you are paying for your VoIP provider you should have means to check the actual situation of voice quality in your VoIP VoIP Network Test HD Voice easily!