MAF ICIMS™ is a reporting and analytics solution for Unified Communication and Collaboration (UC&C) platforms and other data sources such as Session Border Controllers (SBC’s), Gateways, Trading Platforms, Turrents & Dealer Boards. It allows you to gain valuable business and technical insights through its reports, daily dashboards and historical trend monitors. Its flexible, user defined nature means you tell the software what you want to see instead of the software dictating to you what you will see.
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
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
SIP - More than meets the eye
Speakers:
Ofer Cohen - VOIP Group Leader, LivePerson
Yossi Maimon - VOIP Technical Leader, LivePerson
An Introduction to the SIP protocol.
SIP Position in telecommunication networks and the content services.
What is SIP:
The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating sessions consisting of one or several media streams. SIP can be used for two-party (unicast) or multiparty (multicast) sessions. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.
(Source: Wikipedia)
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
Join us for an introductory webinar on VoIP and learn:
- The fundamental principles of VoIP including RTP and SIP
- What voice metrics to measure and why they matter
- The different methods to monitor and troubleshoot VoIP
Comparisons of QoS in VoIP over WIMAX by Varying the Voice codes and Buffer sizeEditor IJCATR
Voice over Internet Protocol (VoIP) is developed for voice communications system based on voice packets transmitted over
IP network with real-time communications of voice across networks using the Internet protocols. Quality of Service (QoS) mechanism
is applied to guarantee successful voice packets transmitted over IP network with reduced delay or drop according to assigned priority
of voice packets. In this paper, the goal of simulation models is present to investigate the performance of VoIP codecs and buffer size
for improving quality of service (QoS) with the simulation results by using OPNET modeler version 14.5. The performance of the
proposed algorithm is analyzed and compared the quality of service for VoIP. The final simulated result shows that the VoIP service
performance best under G.729 voice encoder scheme and buffer size 256 Kb over WiMAX network.
Voice over Internet Protocol with Novel Applicationsirjes
Internet Telephony, often denoted as Voice-over-Internet-Protocol (VoIP), has gained more and
more attention world-wide during the last decades. Voice over Internet Protocol (VoIP) technology has become
a communication alternative with the continuous increasing of Internet bandwidth and rapid advancement of
peer-to-peer (P2P) applications.In this paper three types of VOIP are explained: (i) PC to PC : this is the easiest
way to use VOIP, which enables you to talk and communicate Voice over Internet with all people over the
world. (ii) PC to Phone: which need a gateway that connects IP Network to phone Network. Its uses a device
called an ATA (Analogue Telephone Adaptor).The ATA allows you to connect a standard phone to your
computer or your Internet connection for use with VOIP. The ATA is an analogue to digital converter. (iii)
Phone to Phone: Where you need more gateways that connect IP network to phone networks, more phone
networks that connect Telephone set to gateway, and IP network that connect gateway to gateway. This paper
also explains Internet Protocol (IP) that VOIP uses to transmit voice as packets over an IP network as follows:
H.323 Protocols that provides the technical requirements for voice communication over LANs, while assuming
that Quality of Service isn't provided by LANs. Session Initiation Protocol (SIP) standard which is the standard
for establishing VOIP connections. This paper also explores Voice XML which is a markup language derived
from XML for writing telephone-based speech applications.In this paper also VoiceXML developed
environments(Gateways) are explained, where a list of all VoiceXML developers is included, and the most
popular VoiceXML development environments (Gateways) are explained in details.Finally, in this paper ten
VoiceXML applications are developed
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.
A distributed ip based telecommunication system using sipIJCNCJournal
Voice over Internet Protocol (VoIP) technologies are integral to modern telecommunications because of
their advanced features, flexibility, and economic benefits. Internet Service Providers initially promoted
these technologies by providing low cost local and international calling. At present, there is also a great
deal of interest in using IP-based technologies to replace traditional small and large office telephone
systems that use traditional PBX’s (Private Branch eXchange). Unfortunately, the large majority of the
emerging VoIP based office telephone systems have followed the centralized design of traditional public
and private telephone systems in which all the intelligence in the system is at the core, with quite expensive
hardware and software components and appropriate redundancy for adequate levels of reliability. In this
paper, it is argued that a centralized model for an IP-based telecommunications system fails to exploit the
full capabilities of Internet-inspired communications and that, very simple, inexpensive, elegant and
flexible solutions are possible by deliberately avoiding the centralized approach. This paper describes the
design, philosophy and implementation of a prototype for a fully distributed IP-based Telecommunication
System (IPTS) that provides the essential feature set for office and home telecommunications, including IPbased
long-distance and local calling, and with the support for video as well as data and text. The
prototype system was implemented with an Internet-inspired distributed design using open source software,
with appropriate customizations and configurations.
SIP - More than meets the eye
Speakers:
Ofer Cohen - VOIP Group Leader, LivePerson
Yossi Maimon - VOIP Technical Leader, LivePerson
An Introduction to the SIP protocol.
SIP Position in telecommunication networks and the content services.
What is SIP:
The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating sessions consisting of one or several media streams. SIP can be used for two-party (unicast) or multiparty (multicast) sessions. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.
(Source: Wikipedia)
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
Join us for an introductory webinar on VoIP and learn:
- The fundamental principles of VoIP including RTP and SIP
- What voice metrics to measure and why they matter
- The different methods to monitor and troubleshoot VoIP
Comparisons of QoS in VoIP over WIMAX by Varying the Voice codes and Buffer sizeEditor IJCATR
Voice over Internet Protocol (VoIP) is developed for voice communications system based on voice packets transmitted over
IP network with real-time communications of voice across networks using the Internet protocols. Quality of Service (QoS) mechanism
is applied to guarantee successful voice packets transmitted over IP network with reduced delay or drop according to assigned priority
of voice packets. In this paper, the goal of simulation models is present to investigate the performance of VoIP codecs and buffer size
for improving quality of service (QoS) with the simulation results by using OPNET modeler version 14.5. The performance of the
proposed algorithm is analyzed and compared the quality of service for VoIP. The final simulated result shows that the VoIP service
performance best under G.729 voice encoder scheme and buffer size 256 Kb over WiMAX network.
Voice over Internet Protocol with Novel Applicationsirjes
Internet Telephony, often denoted as Voice-over-Internet-Protocol (VoIP), has gained more and
more attention world-wide during the last decades. Voice over Internet Protocol (VoIP) technology has become
a communication alternative with the continuous increasing of Internet bandwidth and rapid advancement of
peer-to-peer (P2P) applications.In this paper three types of VOIP are explained: (i) PC to PC : this is the easiest
way to use VOIP, which enables you to talk and communicate Voice over Internet with all people over the
world. (ii) PC to Phone: which need a gateway that connects IP Network to phone Network. Its uses a device
called an ATA (Analogue Telephone Adaptor).The ATA allows you to connect a standard phone to your
computer or your Internet connection for use with VOIP. The ATA is an analogue to digital converter. (iii)
Phone to Phone: Where you need more gateways that connect IP network to phone networks, more phone
networks that connect Telephone set to gateway, and IP network that connect gateway to gateway. This paper
also explains Internet Protocol (IP) that VOIP uses to transmit voice as packets over an IP network as follows:
H.323 Protocols that provides the technical requirements for voice communication over LANs, while assuming
that Quality of Service isn't provided by LANs. Session Initiation Protocol (SIP) standard which is the standard
for establishing VOIP connections. This paper also explores Voice XML which is a markup language derived
from XML for writing telephone-based speech applications.In this paper also VoiceXML developed
environments(Gateways) are explained, where a list of all VoiceXML developers is included, and the most
popular VoiceXML development environments (Gateways) are explained in details.Finally, in this paper ten
VoiceXML applications are developed
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.
A distributed ip based telecommunication system using sipIJCNCJournal
Voice over Internet Protocol (VoIP) technologies are integral to modern telecommunications because of
their advanced features, flexibility, and economic benefits. Internet Service Providers initially promoted
these technologies by providing low cost local and international calling. At present, there is also a great
deal of interest in using IP-based technologies to replace traditional small and large office telephone
systems that use traditional PBX’s (Private Branch eXchange). Unfortunately, the large majority of the
emerging VoIP based office telephone systems have followed the centralized design of traditional public
and private telephone systems in which all the intelligence in the system is at the core, with quite expensive
hardware and software components and appropriate redundancy for adequate levels of reliability. In this
paper, it is argued that a centralized model for an IP-based telecommunications system fails to exploit the
full capabilities of Internet-inspired communications and that, very simple, inexpensive, elegant and
flexible solutions are possible by deliberately avoiding the centralized approach. This paper describes the
design, philosophy and implementation of a prototype for a fully distributed IP-based Telecommunication
System (IPTS) that provides the essential feature set for office and home telecommunications, including IPbased
long-distance and local calling, and with the support for video as well as data and text. The
prototype system was implemented with an Internet-inspired distributed design using open source software,
with appropriate customizations and configurations.
Comparative Study for Performance Analysis of VOIP Codecs Over WLAN in Nonmob...Zac Darcy
Voice over IP (VoIP) applications such as Skype, Google Talk, and FaceTime are promising technologies for providing cheaper voice calls to end users over extant networks. Wireless networks such as WiMAX and Wi-Fi focus on providing perfection of service for VoIP. However, there are numerous aspects that affect quality of voice connections over wireless networks [13]. The adoption of Voice over Wireless Local Area Network is on tremendous increase due its relief, non-invasive, economicexpansion, low maintenance cost, universal coverage and basic roaming capabilities.
COMPARATIVE STUDY FOR PERFORMANCE ANALYSIS OF VOIP CODECS OVER WLAN IN NONMOB...Zac Darcy
Voice over IP (VoIP) applications such as Skype, Google Talk, and FaceTime are promising technologies for providing cheaper voice calls to end users over extant networks. ireless networks such as WiMAX and Wi-Fi focus on providing perfection of service for VoIP. However, there are numerous aspects that affect quality of voice connections over wireless networks [13]. The adoption of Voice over Wireless Local Area
Network is on tremendous increase due its relief, non-invasive, economicexpansion, low maintenance cost, universal coverage and basic roaming capabilities. However, expansion Voice over Internet Protocol (VoIP) over Wireless Local Area Network (WLAN) is a challenging task for many network specialist and engineers. Voice codec is one of the most critical components of a VoIP system. In this project, we evaluate
the performance analysis of various codecs such as G.711, G.723 and G.729 over Wi-Fi networks. NS2 WiFi
simulation models are designed. Performance metrics such as Mean Opinion Score (MOS), average
end-to-end latency, and disconcert are evaluated and discussed [13]. 1. In this paper, our area of interest is to compare and study the performance analysis of VoIP codecs in Non-mobility scenarios by changing some parameters and plotting the graphs throughput, End to end Delay, MOS, Packet delivery Ratio, and Jitter by using Network Simulator version.
2. In this paper we analyze the different performance parameters, Recent research has focused on simulation studies with non- mobility scenarios to analyze different VoIP codecs with nodes up to 5. We have simulated the different VoIP codecs in non-mobility scenario with nodes up to 300.
Comparative Study for Performance Analysis of VOIP Codecs Over WLAN in Nonmob...Zac Darcy
Voice over IP (VoIP) applications such as Skype, Google Talk, and FaceTime are promising technologies
for providing cheaper voice calls to end users over extant networks. Wireless networks such as WiMAX and
Wi-Fi focus on providing perfection of service for VoIP. However, there are numerous aspects that affect
quality of voice connections over wireless networks [13]. The adoption of Voice over Wireless Local Area
Network is on tremendous increase due its relief, non-invasive, economicexpansion, low maintenance cost,
universal coverage and basic roaming capabilities. However, expansion Voice over Internet Protocol
(VoIP) over Wireless Local Area Network (WLAN) is a challenging task for many network specialist and
engineers. Voice codec is one of the most critical components of a VoIP system. In this project, we evaluate
the performance analysis of various codecs such as G.711, G.723 and G.729 over Wi-Fi networks. NS2 WiFi simulation models are designed. Performance metrics such as Mean Opinion Score (MOS), average
end-to-end latency, and disconcert are evaluated and discussed [13].
1. In this paper, our area of interest is to compare and study the performance analysis of VoIP
codecs in Non-mobility scenarios by changing some parameters and plotting the graphs
throughput, End to end Delay, MOS, Packet delivery Ratio, and Jitter by using Network
Simulator version.
2. In this paper we analyze the different performance parameters, Recent research has focused on
simulation studies with non- mobility scenarios to analyze different VoIP codecs with nodes up to
5. We have simulated the different VoIP codecs in non-mobility scenario with nodes up to 300.
COMPARATIVE STUDY FOR PERFORMANCE ANALYSIS OF VOIP CODECS OVER WLAN IN NONMOB...Zac Darcy
The recent Voice over IP (VOIP) applications such as Skype, Google Talk, and Face Time have
changed the way people communicate to each other. Due to the low cost, people find VOIP as an
alternative to the expensive traditional Public Switched Telephone Network (PSTN). VOIP has
set of parameters that defined its Quality of Service (QoS) such as end to delay, jitter, packets
loss, Mean Opinion Score (MOS, and throughput[13]. The existing wireless networks such as WiFi offer flexibility to support such applications. At the time the IEEE 802.11 (Wi-Fi) technology
showed great success as cheap wireless internet access. The Motive of this survey paper is to
analyse of Qos in VOIP [13].
COMPARATIVE STUDY FOR PERFORMANCE ANALYSIS OF VOIP CODECS OVER WLAN IN NONMOB...Zac Darcy
Voice over IP (VoIP) applications such as Skype, Google Talk, and FaceTime are promising technologies
for providing cheaper voice calls to end users over extant networks. Wireless networks such as WiMAX and
Wi-Fi focus on providing perfection of service for VoIP. However, there are numerous aspects that affect
quality of voice connections over wireless networks [13]. The adoption of Voice over Wireless Local Area
Network is on tremendous increase due its relief, non-invasive, economicexpansion, low maintenance cost,
universal coverage and basic roaming capabilities. However, expansion Voice over Internet Protocol
(VoIP) over Wireless Local Area Network (WLAN) is a challenging task for many network specialist and
engineers. Voice codec is one of the most critical components of a VoIP system. In this project, we evaluate
the performance analysis of various codecs such as G.711, G.723 and G.729 over Wi-Fi networks. NS2 WiFi simulation models are designed. Performance metrics such as Mean Opinion Score (MOS), average
end-to-end latency, and disconcert are evaluated and discussed [13]
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
CASE STUDY FOR PERFORMANCE ANALYSIS OF VOIP CODECS IN NON-MOBILITY SCENARIOSijcsity
IEEE 802.11 is the most popular standard for WLAN networks. It offers different physical transmission
rates. This paper focuses on this multi transmission rate of 802.11 WLANs and its effect on speech quality.
In non-adaptive systems, when the physical layer switches from a higher transmission rate to a lower one,
different than the one that the VoIP flow needs, the switching may result in congestion, high delay and
packet loss, and consequently speech quality degradation. However, there are some algorithms that adapt
the transmission parameters according to the channel conditions. In this study we demonstrate how
choosing parameter (different codec and packet size) can affect the voice quality, network delay and packet
loss. Further, this study presents a comparison between adaptive and non-adaptive methods. The adaptive method has also been evaluated for different congestion level from perceived speech quality point of view.
As presented at ITExpo 2017 and the April Peerlyst Tel-Aviv security Meetup.
Can your company afford to ignore VoIP security? With the number of attacks on your telephone services and mobile devices your chance of being attacked and financial liability is at an all time high. This session offers an introductory primer to securing your VoIP PBX. This talk will include explanations about common attacks, how they can find you, and common techniques you can use to defend your company.
A NEW SYSTEM ON CHIP RECONFIGURABLE GATEWAY ARCHITECTURE FOR VOICE OVER INTER...csandit
The aim of this paper is to present a new System on Chip (SoC) reconfigurable gateway
architecture for Voice over Internet Telephony (VOIP). Our motivation behind this work is
justified by the following arguments: most of VOIP solutions proposed in the market are based
on the use of a general purpose processor and a DSP circuit. In these solutions, the use of the
serial multiply accumulate circuit is very limiting for the signal processing. Also, in embedded
VOIP based DSP applications, the DSP works without MMU (memory management unit). This
is a serious limitation because VOIP solutions are multi-task based. In order to overcome these
problems, we propose a new VOIP gateway architecture built around the OpenRisc-1200-V3
processor. This last one integrates a DSP circuit as well as a MMU. The hardware architecture
is mapped into the VIRTEX-5 FPGA device. We propose a design methodology based on the
design for reuse and design with reuse concepts. We demonstrate that the proposed SoC
architecture is reconfigurable, scalable and the final RTL code can be reused for any FPGA or
ASIC technology. Performances measures, in the VIRTEX-5 FPGA device family, show that the
SOC-gateway architecture occupies 52% of the FPGA in term of slice LUT, 42% of IOBs, 60%
of bloc memory, 8% of integrated DSP, 16% of PLL and the total power is estimated at
4.3Watts.
MAF NMS Number Management System for DiD-DDi range management - a deeper lookMAF InfoCom
Tracking and managing employee Direct-in-Dial (DID or DDI depending on which side of the pond you are on) and extensions for (IP)PBX & Unfied Communications can be a costly and time consuming task for companies of any size. Not anymore with MAF NMS. It assists companies in managing DID’s and Extensions which results in reduced costs and improved business processes.
MAF ICIMS™ - microsoft teams real time agent precence and call queues reportingMAF InfoCom
MAF ICIMS™ is a reporting and analytics solution for Unified Communication and Collaboration (UC&C) platforms and other data sources such as Session Border Controllers (SBC’s), Gateways, Trading Platforms, Turrents & Dealer Boards. It allows you to gain valuable business and technical insights through its reports, daily dashboards and historical trend monitors. Its flexible, user defined nature means you tell the software what you want to see instead of the software dictating to you what you will see.
MAF ICIMS™ Monitoring, Analytics & Reporting for Microsoft Teams and UC - res...MAF InfoCom
MAF ICIMS™ is a reporting and analytics solution for Unified Communication and Collaboration (UC&C) platforms and other data sources such as Session Border Controllers (SBC’s), Gateways, Trading Platforms, Turrents & Dealer Boards. It allows you to gain valuable business and technical insights through its reports, daily dashboards and historical trend monitors. Its flexible, user defined nature means you tell the software what you want to see instead of the software dictating to you what you will see.
MAF ICIMS™ Monitoring, Analytics & Reporting for Microsoft Teams and UC - rep...MAF InfoCom
MAF ICIMS™ is a reporting and analytics solution for Unified Communication and Collaboration (UC&C) platforms and other data sources such as Session Border Controllers (SBC’s), Gateways, Trading Platforms, Turrents & Dealer Boards. It allows you to gain valuable business and technical insights through its reports, daily dashboards and historical trend monitors. Its flexible, user defined nature means you tell the software what you want to see instead of the software dictating to you what you will see.
MAF ICIMS™ Monitoring, Analytics & Reporting for Microsoft Teams and UC - Das...MAF InfoCom
MAF ICIMS™ is a reporting and analytics solution for Unified Communication and Collaboration (UC&C) platforms and other data sources such as Session Border Controllers (SBC’s), Gateways, Trading Platforms, Turrents & Dealer Boards. It allows you to gain valuable business and technical insights through its reports, daily dashboards and historical trend monitors. Its flexible, user defined nature means you tell the software what you want to see instead of the software dictating to you what you will see
MAF DMS™ Device Management System - Supported DevicesMAF InfoCom
MAF DMS™ Device Management System is an additional module to MAF ICIMS™ providing inventory details, management and quality reporting for Epos, Jabra and Poly headsets. It allows you to realise productivity gains whilst driving the user adoption of devices. The integration with MAF ICIMS™ allows end-to-end visibility of both headset and UC&C call quality with support for Microsoft Teams, Skype for Business, Amazon Chime and Connect and Cisco.
MAF DMS™ Device Management - A Deeper LookMAF InfoCom
MAF DMS™ is an additional module to MAF ICIMS™ providing inventory details, management and quality reporting for Epos, Jabra and Poly headsets. It allows you to realise productivity gains whilst driving the user adoption of devices. The integration with MAF ICIMS™ allows end-to-end visibility of both headset and UC&C call quality with support for Microsoft Teams, Skype for Business, Amazon Chime and Connect and Cisco.
In 2015, I used to write extensions for Joomla, WordPress, phpBB3, etc and I ...Juraj Vysvader
In 2015, I used to write extensions for Joomla, WordPress, phpBB3, etc and I didn't get rich from it but it did have 63K downloads (powered possible tens of thousands of websites).
Understanding Globus Data Transfers with NetSageGlobus
NetSage is an open privacy-aware network measurement, analysis, and visualization service designed to help end-users visualize and reason about large data transfers. NetSage traditionally has used a combination of passive measurements, including SNMP and flow data, as well as active measurements, mainly perfSONAR, to provide longitudinal network performance data visualization. It has been deployed by dozens of networks world wide, and is supported domestically by the Engagement and Performance Operations Center (EPOC), NSF #2328479. We have recently expanded the NetSage data sources to include logs for Globus data transfers, following the same privacy-preserving approach as for Flow data. Using the logs for the Texas Advanced Computing Center (TACC) as an example, this talk will walk through several different example use cases that NetSage can answer, including: Who is using Globus to share data with my institution, and what kind of performance are they able to achieve? How many transfers has Globus supported for us? Which sites are we sharing the most data with, and how is that changing over time? How is my site using Globus to move data internally, and what kind of performance do we see for those transfers? What percentage of data transfers at my institution used Globus, and how did the overall data transfer performance compare to the Globus users?
Custom Healthcare Software for Managing Chronic Conditions and Remote Patient...Mind IT Systems
Healthcare providers often struggle with the complexities of chronic conditions and remote patient monitoring, as each patient requires personalized care and ongoing monitoring. Off-the-shelf solutions may not meet these diverse needs, leading to inefficiencies and gaps in care. It’s here, custom healthcare software offers a tailored solution, ensuring improved care and effectiveness.
Listen to the keynote address and hear about the latest developments from Rachana Ananthakrishnan and Ian Foster who review the updates to the Globus Platform and Service, and the relevance of Globus to the scientific community as an automation platform to accelerate scientific discovery.
Providing Globus Services to Users of JASMIN for Environmental Data AnalysisGlobus
JASMIN is the UK’s high-performance data analysis platform for environmental science, operated by STFC on behalf of the UK Natural Environment Research Council (NERC). In addition to its role in hosting the CEDA Archive (NERC’s long-term repository for climate, atmospheric science & Earth observation data in the UK), JASMIN provides a collaborative platform to a community of around 2,000 scientists in the UK and beyond, providing nearly 400 environmental science projects with working space, compute resources and tools to facilitate their work. High-performance data transfer into and out of JASMIN has always been a key feature, with many scientists bringing model outputs from supercomputers elsewhere in the UK, to analyse against observational or other model data in the CEDA Archive. A growing number of JASMIN users are now realising the benefits of using the Globus service to provide reliable and efficient data movement and other tasks in this and other contexts. Further use cases involve long-distance (intercontinental) transfers to and from JASMIN, and collecting results from a mobile atmospheric radar system, pushing data to JASMIN via a lightweight Globus deployment. We provide details of how Globus fits into our current infrastructure, our experience of the recent migration to GCSv5.4, and of our interest in developing use of the wider ecosystem of Globus services for the benefit of our user community.
Prosigns: Transforming Business with Tailored Technology SolutionsProsigns
Unlocking Business Potential: Tailored Technology Solutions by Prosigns
Discover how Prosigns, a leading technology solutions provider, partners with businesses to drive innovation and success. Our presentation showcases our comprehensive range of services, including custom software development, web and mobile app development, AI & ML solutions, blockchain integration, DevOps services, and Microsoft Dynamics 365 support.
Custom Software Development: Prosigns specializes in creating bespoke software solutions that cater to your unique business needs. Our team of experts works closely with you to understand your requirements and deliver tailor-made software that enhances efficiency and drives growth.
Web and Mobile App Development: From responsive websites to intuitive mobile applications, Prosigns develops cutting-edge solutions that engage users and deliver seamless experiences across devices.
AI & ML Solutions: Harnessing the power of Artificial Intelligence and Machine Learning, Prosigns provides smart solutions that automate processes, provide valuable insights, and drive informed decision-making.
Blockchain Integration: Prosigns offers comprehensive blockchain solutions, including development, integration, and consulting services, enabling businesses to leverage blockchain technology for enhanced security, transparency, and efficiency.
DevOps Services: Prosigns' DevOps services streamline development and operations processes, ensuring faster and more reliable software delivery through automation and continuous integration.
Microsoft Dynamics 365 Support: Prosigns provides comprehensive support and maintenance services for Microsoft Dynamics 365, ensuring your system is always up-to-date, secure, and running smoothly.
Learn how our collaborative approach and dedication to excellence help businesses achieve their goals and stay ahead in today's digital landscape. From concept to deployment, Prosigns is your trusted partner for transforming ideas into reality and unlocking the full potential of your business.
Join us on a journey of innovation and growth. Let's partner for success with Prosigns.
Code reviews are vital for ensuring good code quality. They serve as one of our last lines of defense against bugs and subpar code reaching production.
Yet, they often turn into annoying tasks riddled with frustration, hostility, unclear feedback and lack of standards. How can we improve this crucial process?
In this session we will cover:
- The Art of Effective Code Reviews
- Streamlining the Review Process
- Elevating Reviews with Automated Tools
By the end of this presentation, you'll have the knowledge on how to organize and improve your code review proces
Innovating Inference - Remote Triggering of Large Language Models on HPC Clus...Globus
Large Language Models (LLMs) are currently the center of attention in the tech world, particularly for their potential to advance research. In this presentation, we'll explore a straightforward and effective method for quickly initiating inference runs on supercomputers using the vLLM tool with Globus Compute, specifically on the Polaris system at ALCF. We'll begin by briefly discussing the popularity and applications of LLMs in various fields. Following this, we will introduce the vLLM tool, and explain how it integrates with Globus Compute to efficiently manage LLM operations on Polaris. Attendees will learn the practical aspects of setting up and remotely triggering LLMs from local machines, focusing on ease of use and efficiency. This talk is ideal for researchers and practitioners looking to leverage the power of LLMs in their work, offering a clear guide to harnessing supercomputing resources for quick and effective LLM inference.
Gamify Your Mind; The Secret Sauce to Delivering Success, Continuously Improv...Shahin Sheidaei
Games are powerful teaching tools, fostering hands-on engagement and fun. But they require careful consideration to succeed. Join me to explore factors in running and selecting games, ensuring they serve as effective teaching tools. Learn to maintain focus on learning objectives while playing, and how to measure the ROI of gaming in education. Discover strategies for pitching gaming to leadership. This session offers insights, tips, and examples for coaches, team leads, and enterprise leaders seeking to teach from simple to complex concepts.
OpenFOAM solver for Helmholtz equation, helmholtzFoam / helmholtzBubbleFoamtakuyayamamoto1800
In this slide, we show the simulation example and the way to compile this solver.
In this solver, the Helmholtz equation can be solved by helmholtzFoam. Also, the Helmholtz equation with uniformly dispersed bubbles can be simulated by helmholtzBubbleFoam.
May Marketo Masterclass, London MUG May 22 2024.pdfAdele Miller
Can't make Adobe Summit in Vegas? No sweat because the EMEA Marketo Engage Champions are coming to London to share their Summit sessions, insights and more!
This is a MUG with a twist you don't want to miss.
Navigating the Metaverse: A Journey into Virtual Evolution"Donna Lenk
Join us for an exploration of the Metaverse's evolution, where innovation meets imagination. Discover new dimensions of virtual events, engage with thought-provoking discussions, and witness the transformative power of digital realms."
Climate Science Flows: Enabling Petabyte-Scale Climate Analysis with the Eart...Globus
The Earth System Grid Federation (ESGF) is a global network of data servers that archives and distributes the planet’s largest collection of Earth system model output for thousands of climate and environmental scientists worldwide. Many of these petabyte-scale data archives are located in proximity to large high-performance computing (HPC) or cloud computing resources, but the primary workflow for data users consists of transferring data, and applying computations on a different system. As a part of the ESGF 2.0 US project (funded by the United States Department of Energy Office of Science), we developed pre-defined data workflows, which can be run on-demand, capable of applying many data reduction and data analysis to the large ESGF data archives, transferring only the resultant analysis (ex. visualizations, smaller data files). In this talk, we will showcase a few of these workflows, highlighting how Globus Flows can be used for petabyte-scale climate analysis.
Experience our free, in-depth three-part Tendenci Platform Corporate Membership Management workshop series! In Session 1 on May 14th, 2024, we began with an Introduction and Setup, mastering the configuration of your Corporate Membership Module settings to establish membership types, applications, and more. Then, on May 16th, 2024, in Session 2, we focused on binding individual members to a Corporate Membership and Corporate Reps, teaching you how to add individual members and assign Corporate Representatives to manage dues, renewals, and associated members. Finally, on May 28th, 2024, in Session 3, we covered questions and concerns, addressing any queries or issues you may have.
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Enterprise Resource Planning System includes various modules that reduce any business's workload. Additionally, it organizes the workflows, which drives towards enhancing productivity. Here are a detailed explanation of the ERP modules. Going through the points will help you understand how the software is changing the work dynamics.
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Venez le découvrir lors de cette session ignite
2. www.mafinfo.com info@mafinfo.com
Summary
Report Designer
Rates
Carrier The name of the company which provides telecommunication services.
Cost Cost of the services provided by telecommunication entity.
Cost 2 Cost of the services provided by telecommunication entity.
Destination Type Can be set up as International, International-Mobile, Local or can be
personalized based on prefixes.
Call Details
Account …
Call ID A unique identifier for every call.
Call Type Call type name abbreviated
Call Type Name Type of the call like Abandoned, Busy, Conference.
Channel A gateway can have more channels.
Conf. Organizer The sip address of the conference organizer.
Conference ID An identifier which allows you to follow the call chain.
Data Source The set up and configuration method of collecting CDRs from Skype for
Business.
Date Date on which call took place.
Day Day on which call took place.
Dialed Number Depending on context that field can be either the external number
dialing in(CLID) or the number that a user dialed.
Direction Defines weather or not the call was incoming, outgoing or Internal.
Duration Duration of time that call was live from the moment it was picked up and
until it ended.
Extension Extension number.
Extension Type Can be cellular, phone, fax or sip.
Extra string 1 Personalized parameters can be added.
Extra string 2 Personalized parameters can be added.
Extra string 3 Personalized parameters can be added.
Gateway Is a network node that connects two networks using different protocols
together.
Referred by Is a field in CDRs which contains information regarding the call chain,
usually it’s the extension that passed the call via a transfer.
Ring time Total time call rang before connection or disconnection.
Service type This is the call modality (IM / App share / Voice / Video / Data)
Time The Time the call took place.
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Call Types
Is app. Sharing If the users are sharing their screens it will show ‘Y’.
Is Conference ‘Y’ will appear in the column if the call is conference or ‘N’ if it
isn’t.
Is Federated ‘Y’ will appear in the column if the call is federated or ‘N’ if it isn’t.
Is File Transfer ‘Y’ will appear in the column if the call contains file transfers.
Is Response Group ‘Y’ will appear in the column if the call is response group call or
‘N’ if it isn’t.
Destination
Location Where geographically the call came from.
Phone The location based on the dialed number.
Phone Group A group of destinations.
Region Continents.
Employee
Employee Name and sip address of the employee.
Employee first Name Users First name.
Employee ID Users Sip address.
Employee Last Name Users Last Name.
Employee name Users First and Last Name.
Extension location Location of the extension from which the call was made.
Hierarchy
Ancestor Unit In multilevel hierarchy, i.e. Company->Town->Department ->
Employee (ancestor unit of the employee is town and
organization unit is Department)
Organization unit Assigned department within organization
IP Fields
Connection Type The connection type that user is using, i.e. Ethernet, WI-FI, Wired.
Dest. Audio codec Is a codec (a device or computer program capable of encoding
or decoding a digital data stream) that encodes or decodes
audio.
Dest. Resolution …
Dest. Video codec Is an electronic circuit or software that compresses or
decompresses digital video. It converts uncompressed video to a
compressed format or vice versa.
Destination IP Internet Protocol address is a numerical label of the recipient,
assigned to each device connected to a computer network that
uses the Internet Protocol for communication.
Destination IP v6 Is the most recent version of the Internet Protocol (IP), the
communications protocol that provides an identification and
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location system for computers on networks and routes traffic
across the Internet.
From reflexive local IP Reflexive local IP contain condition statements (entries) that
define criteria for permitting IP packets.
Jitter Measures the variability of packet delay and results in a distorted
or choppy audio experience. Jitter can increase latency on
networks.
Packets A network packet is a formatted unit of data carried by a packet-
switched network.
Latency Is a time interval between the stimulation and response, or, from
a more general point of view, a time delay between the cause and
the effect of some physical change in the system being observed.
MOS Mean opinion score – is the gold standard measurement to
gauge the perceived audio quality.
Can be between 1 and 5: - 1 (Bad)
- 2 (Poor)
- 3 (Fair)
- 4 (Good)
- 5 (Excellent)
Octet Is a unit of digital information in computing and
telecommunications that consists of eight bits. The term is often
used when the term byte might be ambiguous, as the byte has
historically been used for storage units of a variety of sizes.
Octets received The amount of received octets.
Octets sent The number of octets sent.
Orig. audio codec Is a codec (a device or computer program capable of encoding
or decoding a digital data stream) that encodes or decodes
audio.
Orig. video codec Is an electronic circuit or software that compresses or
decompresses digital video. It converts uncompressed video to a
compressed format or vice versa.
Originator IP Internet Protocol address is a numerical label of the remitter,
assigned to each device connected to a computer network that
uses the Internet Protocol for communication.
Originator IPv6 Is the most recent version of the Internet Protocol (IP), the
communications protocol that provides an identification and
location system for computers on networks and routes traffic
across the Internet.
Packets lost Packet Loss (%) represents the % of packets that did not make it
to their destination. Packet loss will cause the audio to be
distorted or missing (on the receiver end).
Packets Received The amount of received packets.
Packets sent The amount of sent packets.
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Pool Is a set of resources that are kept ready to use, rather than acquired on
use and released afterwards.
Quality Is an international standard, developed by Virtual Socket Interface
Alliance for measuring IP or SIP (Silicon intellectual property) quality and
examining the practices used to design, integrate and support the SIP.
Server Is a computer program or a device that provides functionality for other
programs or devices, called "clients". These can be either physical or
virtual machines.
SIP response code Is a signaling protocol used for controlling communication sessions such
as Voice over IP telephone calls. SIP is based around request/response
transactions, in a similar manner to the Hypertext Transfer Protocol
(HTTP). Each transaction consists of a SIP request (which will be one of
several request methods), and at least one response.
Subnet Is a logical subdivision of an IP network. The practice of dividing a
network into two or more networks is called subnetting.
Subnet location Location of the subnetwork.
To reflexive local IP Reflexive local IP contain condition statements (entries) that define
criteria for permitting IP packets.
VPN A virtual private network extends a private network across a public
network, and enables users to send and receive data across shared or
public networks as if their computing devices were directly connected to
the private network.
Skype for Business
App. Sh. Avg. jitter Measures the variability of packet delay and results in a distorted or
choppy audio experience. Jitter can increase latency on networks.
Avg. Net MOS Network MOS is a prediction of the wideband Listening Quality Mean
Opinion Score (MOS-LQ) of audio that is played to the user. This value
takes into consideration only network factors such as codec used, packet
loss, packet reorder, packet errors and jitter.
Call admission control Prevents oversubscription of VoIP networks. It is used in the call set-
up phase and applies to real-time media traffic as opposed to data
traffic.
Callee The agent / employee / user receiving a call.
Callee app. Sh. Relative one-way avg. Optimal value for the relative one-way delay between
the two media endpoints involved in the application sharing. This is a
single-hop latency measure.
Callee app. Sharing bandwidth (Kbps) Is a type of resource allocation designed to model the
real-world allocation of bandwidth to many users in a network.
Callee audio bandwidth (Kbps) This refers to the frequency range a device can carry without
degrading any of the information. It’s also used in digital communication
to show how much information something can transfer over a given
time.
6. www.mafinfo.com info@mafinfo.com
Callee audio packets lost rate Packet Loss (%) represents the % of packets that did not make
it to their destination. Packet loss will cause the audio to be
distorted or missing (on the receiver end).
Callee audio round trip Is the most common measure of latency and is measured in ms.
Callee avg. jitter Measures the variability of packet delay and results in a distorted
or choppy audio experience on the receiving end.
Callee avg. listening MOS Is a prediction of the wideband Listening Quality (MOS-LQ)) of
the audio stream that is played to the user.
Callee avg. MOS Average means opinion score.
Callee avg. net MOS degradation Network MOS Degradation for the whole call. This metric
shows the amount the Network MOS was reduced because of
jitter and packet loss.
Callee avg. sending MOS Is a prediction of the wideband Listening Quality Mean Opinion
Score (MOS-LQ) of the audio stream that is being sent from the
user. This value takes into consideration the speech and noise
levels of the user along with any distortions, and from this data
predicts how a large group of users would rate the audio quality
they hear.
Callee client type Client type, i.e. Skype for Windows, Skype for iPhone, Skype for
Android.
Callee client version Client version.
Callee conv. MOS Is a prediction of the narrowband Conversational Quality (MOS-
CQ) of the audio stream that is played to the user. This value takes
into consideration the listening quality of the audio played and
sent across the network, the speech and noise levels for both
audio streams, and echoes. It represents how a large group of
people would rate the quality of the connection for holding a
conversation.
Callee dynamic capability % Percentage of the call where the client experienced high CPU
load when processing video.
Callee echo mic in Echo that was present in the microphone. Typically, you will see
low values for headsets or handsets, and higher values for
speaker phones or stand-alone speakers.
Callee echo send Echo transmitted to other users on the call.
Callee end point Is a device or node that is connected to the LAN or WAN and
accepts communications back and forth across the network.
Callee inbound video frame rate avg. The average video frame rate received during the call.
Callee low frame rate call % Percentage of low frame rate call.
Callee low network BW Is the minimum rate of data transfer across a given path.
Bandwidth may be characterized as network bandwidth, data
bandwidth, or digital bandwidth.
7. www.mafinfo.com info@mafinfo.com
Callee max jitter Measures the maximum variability of packet delay and
results in a distorted or choppy audio experience.
Callee max net MOS degradation This metric shows the maximum amount the Network
MOS that was reduced because of jitter and packet loss.
Callee MIC. not functioning Calls in which the capture device was not functioning at
an acceptable level. A high value suggests that quality
issues with the call were primarily due to the capture
device not working as expected.
Callee min net MOS This metric shows the minimum amount the Network
MOS was reduced because of jitter and packet loss.
Callee near end to echo Used in telephony to improve voice quality by preventing
echo from being created or removing it after it is already
present.
Callee network connection Network connection.
Callee outbound video frame rate avg. The average video frame rate sent during the call.
Callee PAI P-Asserted-Identity.
Callee ratio concealed samples avg. Concealing audio samples is a technique used to deal
with dropped network packets.
Callee RDP tile processing latency avg. Acceptable value of the average RDP tile processing
latency in the AS Conferencing Server over the duration
of the viewing session.
Callee recv. frame rate avg. Average video frame rate used by the receiver
Callee render device Device (for example, a headset or speakers) used for
receiving audio.
Callee spk. not functioning Calls in which the render device was not functioning at an
acceptable level. A high value suggests that quality issues
with the call were primarily due to the render device not
working as expected.
Callee spoiled tile % total Total percentage of spoiled RDP tiles
Callee subnet The subnet the callee resides on.
Callee URI A Uniform Resource Identifier is a string of characters
used to identify a resource.
Callee video avg. jitter Average jitter in video calls.
Callee video bandwidth (Kbps) Video calls bandwidth.
Callee video local frame loss % avg. The percentage of total video frames that are lost.
Callee video packets loss rate The packet loss rate for video calls.
Callee video post FECPLR The packet loss rate after forward error correction has
been applied.
Callee video round trip Round trip time for video calls.
Callee voice switch Calls which had to be placed into half duplex mode. In half
duplex mode, communication can travel in only one
direction at a given time.
8. www.mafinfo.com info@mafinfo.com
Callee VPN A virtual private network extends a private network across a
public network and enables users to send and receive data across
shared or public networks as if their computing devices were
directly connected to the private network.
Caller The agent / Employee / user making a call.
Caller app. Sh. Relative one-way avg. Optimal value for the relative one-way delay between
the two media endpoints involved in the application sharing. This
is a single-hop latency measure.
Caller app. Sharing bandwidth (Kbps) – Is a type of resource allocation designed to model
the real-world allocation of bandwidth to many users in a
network.
Caller audio bandwidth (Kbps) This refers to the frequency range a device can carry without
degrading any of the information. It’s also used in digital
communication to show how much information something can
transfer over a given time.
Caller audio packets lost rate Packet Loss represents the number of packets that did not
make it to their intended destination. Packet loss will cause the
audio to be distorted or missing.
Caller audio round trip Is the most common measure of latency and is measured in MS.
Caller avg. jitter Measures the variability of packet delay and results in a distorted
or choppy audio experience.
Caller avg. listening MOS Is a prediction of the wideband Listening Quality (MOS-LQ)) of
the audio stream that is played to the user.
Caller avg. MOS Average Means Opinion Score.
Caller avg. net MOS degradation Average network MOS degradation is an integer represents
the amount of the MOS value lost to network affects.
Caller avg. sending MOS Is a prediction of the wideband Listening Quality Mean Opinion
Score (MOS-LQ) of the audio stream that is being sent from the
user. This value takes into consideration the speech and noise
levels of the user along with any distortions, and from this data
predicts how a large group of users would rate the audio quality
they hear.
Caller capture device The Microphone or recording device use to capture audio.
Caller client type Client type, i.e. Skype for Windows, Skype for iPhone, Skype for
Android.
Caller client version Client version.
Caller conv. MOS Is a prediction of the narrowband Conversational Quality (MOS-
CQ) of the audio stream that is played to the user. This value takes
into consideration the listening quality of the audio played and
sent across the network, the speech and noise levels for both
audio streams, and echoes. It represents how a large group of
people would rate the quality of the connection for holding a
conversation.
9. www.mafinfo.com info@mafinfo.com
Caller dynamic capability % Percentage of the call where the client experienced high
CPU load when processing video.
Caller echo mic in Echo that was present in the microphone.
Caller echo send Echo transmitted to other users on the call.
Caller end point Is a device or node that is connected to the LAN or WAN
and accepts communications back and forth across the
network.
Caller inbound video frame rate avg. The average video frame rate sent during the call.
Caller low frame rate call % Percentage of low frame rate within a call.
Caller low network BW Is the minimum rate of data transfer across a given path.
Bandwidth may be characterized as network bandwidth,
data bandwidth, or digital bandwidth.
Caller max jitter Measures the maximum variability of packet delay and
results in a distorted or choppy audio experience.
Caller max net MOS degradation This metric shows the maximum amount the Network
MOS that was reduced because of jitter and packet loss.
Caller MIC. not functioning Calls in which the capture device was not functioning at
an acceptable level. A high value suggests that quality
issues with the call were primarily due to the capture
device not working as expected.
Caller min net MOS This metric shows the minimum amount the Network
MOS was reduced because of jitter and packet loss.
Caller near end to echo Used in telephony to improve voice quality by preventing
echo from being created or removing it after it is already
present.
Caller network connection Shows the network the caller connected to Wired / WIFI /
ethernet Etc.
Caller outbound video frame rate avg. The average video frame rate sent during the call
Caller PAI P-Asserted-Identity.
Caller ratio concealed samples avg. Concealing audio samples is a technique used to deal
with dropped network packets.
Caller RDP tile processing latency avg. Acceptable value of the average RDP tile processing
latency in the AS Conferencing Server over the duration
of the viewing session.
Caller recv. frame rate avg. Average video frame rate used by the receiver.
Caller render device Device (for example, a headset or speakers) used for
receiving audio.
Caller spk. not functioning Calls in which the render device was not functioning at an
acceptable level. A high value suggests that quality issues
with the call were primarily due to the render device not
working as expected.
Caller spoiled tile % total Total percentage of spoiled RDP tiles.
Caller subnet The subnet the caller resides on.
10. www.mafinfo.com info@mafinfo.com
Caller URI A Uniform Resource Identifier is a string of characters used to
identify a resource.
Caller video avg. jitter Average jitter in video calls.
Caller video bandwidth (Kbps) Video calls bandwidth.
Caller video local frame loss % avg. The percentage of total video frames that are lost.
Caller video packets loss rate Packet Loss represents the number of packets that did not
make it to their intended destination. Packet loss will cause the
audio to be distorted or missing.
Caller video post FECPLR The packet loss rate after forward error correction has been
applied.
Caller video round trip This measure the average round-trip time for RTP packets
between endpoints. When the latency is high, users are likely to
hear a delay in the audio
Caller voice switch Calls which had to be placed into half duplex mode. In half duplex
mode, communication can travel in only one direction at a given
time.
Caller VPN A virtual private network extends a private network across a
public network and enables users to send and receive data across
shared or public networks as if their computing devices were
directly connected to the private network.
Client alias An alias is an alternate name that can be used to make a
connection. The alias encapsulates the required elements of a
connection string and exposes them with a name chosen by the
user.
Client version Version of the client.
Diagnostic ID Is a unique identifier (in the form of an ms-diagnostics header)
that gets attached to a SIP message, while the Diagnostic header
provides an accompanying description for the Diagnostic ID.
Disconnected by phone The connection was interrupted due to phone issues.
Disconnected by user The connection was interrupted due to user issues.
Error category Type of the error occurred.
Error description Description of the error with details.
Extension client type Client type that the extension is using.
HD quality High definition quality.
NMOS degradation (jitter) Network MOS Degradation for the complete call. This metric
shows the amount the Network MOS was reduced because of
jitter.
NMOS degradation (packet loss) Network MOS Degradation for the complete call. This
metric shows the amount the Network MOS was reduced
because of packet loss.
Pool Is a set of resources that are kept ready to use, rather than
acquired on use and released afterwards.
Rating …
11. www.mafinfo.com info@mafinfo.com
Rating categories …
Ratio compressed samples avg. Quantify the reduction in data-representation size
produced by a data compression algorithm.
Ratio stretched samples avg. …
SD quality Standard quality.
Server Is a computer program or a device that provides
functionality for other programs or devices, called
"clients"
Video allocated bandwidth The amount of bandwidth that is allocated for video calls.
Video resolution Resolution of video calls.
Response Group
Queue name Name of the queue
Response group description Description of the Response group
SIP address Email address used to configure the Skype for Business
account.
Telephone Telephone.
Summary
Callee NMOS degradation* Network MOS Degradation for the whole call. This metric
shows the amount the Network MOS was reduced
because of jitter and packet loss
Caller NMOS degradation * Network MOS Degradation for the whole call. This metric
shows the amount the Network MOS was reduced
because of jitter and packet loss
Calls* Outbound (out to PSTN)/ inbound (incoming from PSTN)
/ internal calls (internal call between Skype for business
users)
Cost * Rate associated with making or receiving calls.
Cost 2 * Rate associated with making or receiving calls.
Duration * Total time call was live. Picked up (connected) -> hung up
(disconnected)
Extensions * Extension number.
Jitter * Measures the variability of packet delay and results in a
distorted or choppy audio experience. Jitter can increase
latency on networks.
MOS * Mean opinion score – is the gold standard measurement
to gauge the perceived audio quality.
Can be between 1 and 5: - 1 (Bad)
- 2 (Poor)
- 3 (Fair)
- 4 (Good)
- 5 (Excellent)
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Ring time * Total time call range for.
Time * Time at which the call took place.
Report Builder
General
Date Can be set for a specific day or range.
Time Can be set for a specific time or range.
Duration The elapsed time between answer and disconnect
Ring Time Total time call rang before connection or disconnection.
Cost Cost of the services provided by telecommunication entity.
Direction Defines weather or not the call was incoming, outgoing or
Internal.
Incoming A call that is coming into the organization
Outbound An outgoing call from a user
Internal A call that is between users within the same organization
Service type This is the call modality (IM / App share / Voice / Video / Data)
Voice Audio calls.
Video Video calls.
App. Sharing Sharing screen during a Skype call or conference.
IM Instant messages between Skype4b users.
Data Files transferred between Skype users
Call Types
Abandoned The caller hung up the call without being answered. Duration of
the call is 0 and ring time reflects the amount of time the call was
being presented.
Start For an internal call, Skype will generate 2 call detail records
(CDRs), start leg has caller extension in Extension column and
callee extension will be in CLID column.
Transfer Agent picks up calls and transfers it out to another agent or dept.
Conference A service feature that allows a call to be established among three
or more stations in such a manner that each of the stations is able
to communicate with all the other stations.
Pickup CISCO
Tandem – i.e. A call comes outside working hours. The system can be set up to
send the call to an external user or number. In the system it will
appear as one incoming call and one outgoing call.
Presented A call that has rang to an individual agent.
File transfer Files, documents or any data transferred through Skype
Transfer out Transferred call out to PSTN
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Busy The system closed the call depending on the configuration, i.e.
call timeout, lack of voicemail or overflow system.
End For an internal call, Skype will generate 2 call detail records
(CDRs), end leg has callee extension in Extension column and
caller extension in CLID column.
Forward A call which is forwarded to another employee, department etc,
action is done via an automatic system such as response groups.
Response Group Is a feature that lets managers or server administrators route and
queue incoming calls to groups of people, called agents, such as
for a help desk or a customer service desk.
Park Is a feature of some telephone systems that allows a user to put
a call on hold at one telephone set and continue the conversation
from any other telephone set.
Voice mail Is a method of storing voice messages electronically for later
retrieval by intended recipients.
Personal Calls identified for a personal purpose.
Error A call which is identified as inaccurate or incorrect
Scheduled A call set for a specific time.
Federated Enables a Skype for Business user to connect with users in other
organizations that use Skype for Business as well as those that
host their own Skype for Business Server on-premises.
CUCM Call Types
Intercom A dedicated voice service within a specified user environment.
Barge Enables you to drop in on live calls to speak with both the caller
and the agent.
IVR Interactive voice response is a technology that allows a computer
to interact with users through the use of voice and DTMF tones
input via a keypad.
Malicious CISCO
Mobility Calls though a mobile phone.
HandIn CISCO
HandOut CISCO
Cell pick up CISCO
Call Type Abbreviations
A = Abandoned Call
B = Busy Call
X = Transferred Call
F = Forwarded Call
T = Tandem Call
S = Start Leg
E = End Leg
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Err = Error Call
C = Conference Call
H = Hold Call
Pck = Pickup Call
Icom = Intervom Call
M = Mobility Call
MHin = Mobility HandIn
MHout = Mobility HandOut
CPck = Cell Pickup
IVR = IVR Call
Prk = Call Park
Mal = Malicious Call
Brg = Barge Call
Organization Structure
Organization Structure The way in which employees / departments / teams are set up in
AD.
Extension Extension number (cellular / fax / phone /SIP)
Location Where geographically the call came from.
Referred by Is a field in CDRs which contains information regarding the call
chain, usually it’s the extension that passed the call via a transfer.
Employee Name and sip address of the employee .
Department A subset or team of users within the organization.
Response Groups
Response Group Is a feature that lets managers or server administrators route and
queue incoming calls to groups of people, called agents, such as
for a help desk or a customer service desk.
Queue name The name assigned to a group of employees (Response Group)
All legs Shows calls that have been transferred or bounced between
several agents or response groups.
Destination
Dialed number / CLID The number that the user call out to.
Destinations The location/destination on the call, based on phone directory
Directory groups …
Destination Types Can be international, national, international mobile
Gateway
Gateway Is a network node that connects two networks using different
protocols together.
Channel Is a separate path through which signals can flow
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Carriers Company that offers communication services over land-wire,
cable, mobile (cellular), point-to-point microwave, and/or
satellite systems.
IP Fields
Originator IP Internet Protocol address is a numerical label of the remitter,
assigned to each device connected to a computer network that
uses the Internet Protocol for communication.
Destination IP Internet Protocol address is a numerical label of the recipient,
assigned to each device connected to a computer network that
uses the Internet Protocol for communication.
Subnet A subnet (short for "subnetwork") is an identifiably separate part
of an organization's network. Typically, a subnet may represent
all the machines at one geographic location, in one building, or
on the same local area network (LAN).
Subnet locations Location of Subnet.
MOS Mean Opinion Score
Quality UCA uses the MS methodology of rate calls either GOOD or
POOR quality.
Connection Type How the call was connected between the participants. E.g.
Wired
Wi-Fi
Mobile broadband
Tunnel
VPN Virtual Private Network
Sort and Summary Sort and group reports by applying specific filters.
Ancestor Unit In multilevel hierarchy, i.e. Company->Town->Department ->
Employee (ancestor unit of the employee is town and
organization unit is Department)
Call type Call types can be (see list of abbreviations above)
Carrier The name of the company which provides telecommunication
services.
Channel Is a separate path through which signals can flow.
Conference ID Each conference is given an induvial ID which allows you to follow
the call chain.
Conference organizer The sip address of the conference organizer (Agent that arranged
/ scheduled conference.
Cost Cost of the services provided by telecommunication entity.
Data Source Set up and configuration method of collecting Call Detail Records
(CDRs) from Skype for Business
Date The Date on which the activity (IM / Voice / Video / Data) took
place.
Destination IP Internet Protocol address is a numerical label of the recipient,
assigned to each device connected to a computer network that
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uses the Internet Protocol for communication.
Destination type Can be set up as International, International-Mobile, Local or can
be personalized based on prefixes.
Dialed Number The number the user dialed.
Direction Defines weather or not the call was incoming, outgoing or
Internal.
Duration The total time between the call being picked up and
disconnected.
Employee Name and sip address of the user / agent.
Extension Extension number.
Extension location Location of extension geographically.
Extension Type Can be cellular, phone, fax or sip.
Gateway Is a network node that connects two networks using different
protocols together.
Month The calendar month in which the activity took place ( IM / Voice
/ Video / Data / App share)
Organization unit Assigned department within the company.
Originator IP Internet Protocol address is a numerical label of the remitter,
assigned to each device connected to a computer network that
uses the Internet Protocol for communication.
Phone The destination of the call, based on phone directory.
Phone group A group of destinations.
Queue name The name assigned to a group of employees (Response Group)
Referred by Is a field in CDRs which contains information regarding the call
chain, usually it’s the extension that passed the call via a transfer.
Response group Is a feature that lets managers or server administrators route and
queue incoming calls to groups of people, called agents, such as
for a help desk or a customer service desk.
Ring time Total time the call rang for, before being connected or
disconnected.
Service Service type (audio, video, app. sharing, data, IM).
Subnet A subnet (short for "subnetwork") is an identifiably separate part
of an organization's network. Typically, a subnet may represent
all the machines at one geographic location, in one building, or
on the same local area network (LAN).
Subnet location Location of Subnet.
Time The time at which the activity took place (IM / Voice / Video /
Data / App Share).
Week The week at which the activity took place (IM / Voice / Video /
Data / App Share).
Generate Click to produce reports, either in the same web page or a new
one.
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Schedule Report Define a frequency on which you wish the report to be run. For
example, Day, Week, Month, Year. Set the time and who you wish
to deliver to.
Save The ability to save you reports to templates.
Clear Reset the report builder to the default settings
Report Options
Format Select the predefined report formats from the list of bespoke
reports you have designed in the report designer
Currency Select preferred currency
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Who we are
Formed in 2000, MAF InfoCom™ is a leading innovative technology provider with two decades
experience delivering solutions for Unified Communications and Collaboration including
Monitoring, Analytics, Reporting, Recording, Headset & Device Management and DID
Management.
We serve tens of thousands customers around the globe, in a large variety of branches. We
have installations in over 50 countries ranging from SME’s to multi-national global enterprises.
In Europe MAF InfoCom™ is the largest provider of UC reporting solutions.
With the market trend towards Unified Communications and Collaboration we expand our
sales across the globe rapidly. Our solutions work with every major UC&C technology.
Our solutions are offered from the Cloud, On-Premises and Partner Hosted to enable our
customers and partners to choose the best model for their needs.
MAF ICIMS™
UC&C Monitoring Analytics & Reporting
MAF ICIMS CC™
Live Wallboards, Real Time Agent Status
MAF NMS™
Number Management System, DID Range Management
MAF UCR™
UC Voice Recorder
MAF DMS™
Inventory Management for Headset and Devices
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