Prosody S from Aculab is a sophisticated telephony media processing software with fully integrated SIP stack. It is extensively used within contact centre and blue light solutions.
The Polycom SoundPoint IP 450 is a mid-range SIP desktop phone that features Polycom HD Voice for high-quality audio, a high-resolution graphical display, and support for productivity applications through an XML microbrowser. It has a three-line LCD screen, 17 dedicated keys, and 4 soft keys. The phone provides clear transmission, integrated applications, and interoperability with SIP platforms.
The document discusses the SurfUP media processing platform for building voice and video infrastructure applications. It highlights sample applications, system architectures, support for voice and video, integration levels and features, and value propositions of the SurfUP platform. Specifically, it allows processing of voice, video and fax on the same DSP, has direct DSP to network interfaces, is an open platform, supports various applications with the same hardware/software, and enables streaming diagnostics.
This document provides an overview of key concepts related to Voice over IP (VoIP) technology. It defines common VoIP terms and standards, describes how VoIP works by breaking analog voice signals into digital packets, and outlines typical system elements like softswitches, terminals, and gateways. It also discusses media standards, signaling protocols, quality of service measures, fax transmission methods, and various Patton Electronics VoIP products.
The Polycom SoundStation Duo is a conference phone that can be used for both analog and VoIP connections. It provides crystal clear audio quality for conferencing through technologies like Polycom HD Voice. The phone has a large display, web-based administration, and supports a variety of connection and provisioning options to make it flexible and easy to deploy.
The Polycom PVX software is an advanced desktop video conferencing application that delivers high-quality audio, video, and content sharing capabilities to PCs and USB cameras. It utilizes H.264 video and CD-quality audio to provide secure video conferencing with features like content and video sharing. The easy to use application supports standard USB cameras and is compatible with Polycom's conferencing systems and management tools.
1. The document discusses Cisco Call Manager Express which tracks VoIP and POTS components like phones, gateways, and bridges. It configures dial peers for Cisco IP Communicator using SCCP and SJphone using SIP.
2. It describes the basics of VoIP and POTS dial peers in Cisco IOS including destination patterns and session targets. It also covers DTMF relay and session protocols.
3. The document outlines development of a basic SIP softphone client using PortSIP SDK that can register with SIP servers and place/receive audio calls using G.729 codec.
Designing Triple-Play Apps Using DSP Resource BoardsVideoguy
The document discusses the optimal hardware and software architectures for designing triple-play applications using DSP resource boards. It recommends using powerful DSPs with external memory that can handle all media types on a single hardware platform. The software should have an open framework with flexible APIs and remote diagnostics to support new features and algorithms from multiple vendors. The media and control paths should be separate to avoid bottlenecks and reduce host processor load.
Cisco CallManager Express (CME) is a call processing solution that provides VoIP functionality for small to medium sized networks of up to 120 IP phones. It allows connection to the PSTN via analog or digital trunks and supports protocols like Skinny and H.323 for call control. CME is configured on Cisco IOS routers and gateways to provide integrated voice and data services over IP.
The Polycom SoundPoint IP 450 is a mid-range SIP desktop phone that features Polycom HD Voice for high-quality audio, a high-resolution graphical display, and support for productivity applications through an XML microbrowser. It has a three-line LCD screen, 17 dedicated keys, and 4 soft keys. The phone provides clear transmission, integrated applications, and interoperability with SIP platforms.
The document discusses the SurfUP media processing platform for building voice and video infrastructure applications. It highlights sample applications, system architectures, support for voice and video, integration levels and features, and value propositions of the SurfUP platform. Specifically, it allows processing of voice, video and fax on the same DSP, has direct DSP to network interfaces, is an open platform, supports various applications with the same hardware/software, and enables streaming diagnostics.
This document provides an overview of key concepts related to Voice over IP (VoIP) technology. It defines common VoIP terms and standards, describes how VoIP works by breaking analog voice signals into digital packets, and outlines typical system elements like softswitches, terminals, and gateways. It also discusses media standards, signaling protocols, quality of service measures, fax transmission methods, and various Patton Electronics VoIP products.
The Polycom SoundStation Duo is a conference phone that can be used for both analog and VoIP connections. It provides crystal clear audio quality for conferencing through technologies like Polycom HD Voice. The phone has a large display, web-based administration, and supports a variety of connection and provisioning options to make it flexible and easy to deploy.
The Polycom PVX software is an advanced desktop video conferencing application that delivers high-quality audio, video, and content sharing capabilities to PCs and USB cameras. It utilizes H.264 video and CD-quality audio to provide secure video conferencing with features like content and video sharing. The easy to use application supports standard USB cameras and is compatible with Polycom's conferencing systems and management tools.
1. The document discusses Cisco Call Manager Express which tracks VoIP and POTS components like phones, gateways, and bridges. It configures dial peers for Cisco IP Communicator using SCCP and SJphone using SIP.
2. It describes the basics of VoIP and POTS dial peers in Cisco IOS including destination patterns and session targets. It also covers DTMF relay and session protocols.
3. The document outlines development of a basic SIP softphone client using PortSIP SDK that can register with SIP servers and place/receive audio calls using G.729 codec.
Designing Triple-Play Apps Using DSP Resource BoardsVideoguy
The document discusses the optimal hardware and software architectures for designing triple-play applications using DSP resource boards. It recommends using powerful DSPs with external memory that can handle all media types on a single hardware platform. The software should have an open framework with flexible APIs and remote diagnostics to support new features and algorithms from multiple vendors. The media and control paths should be separate to avoid bottlenecks and reduce host processor load.
Cisco CallManager Express (CME) is a call processing solution that provides VoIP functionality for small to medium sized networks of up to 120 IP phones. It allows connection to the PSTN via analog or digital trunks and supports protocols like Skinny and H.323 for call control. CME is configured on Cisco IOS routers and gateways to provide integrated voice and data services over IP.
The document discusses CoreStor, an IP recording solution from Delma that can capture and record IP traffic, including VoIP packets. It describes various methods for capturing IP traffic, such as using span ports, port mirroring, conferencing, or custom gateways. CoreStor is designed to integrate seamlessly into existing systems and provide recording in a single chassis. It supports standard computer hardware and includes replay, administration, and analysis client software.
The document provides an overview of VoIP components, standards, architectures and implementation choices. It discusses key VoIP elements like terminals, packetized voice, soft switches, media servers, gateways, LANs/WANs and standards. It also describes common VoIP architectures for computer-computer, computer-phone, phone-phone communication over the internet. Finally, it outlines VoIP solutions for businesses using VoIP-enabled PBXs, replacing PBXs with softswitches, and hosted PBX solutions.
Audio codes solution for genesys sip contact centerLong Nguyen
CHUYÊN CUNG CẤP THIỆT BỊ VÀ GIẢI PHÁP VOIP
TIME TRUE LIFE TECHNOLOGY JOINT STOCK COMPANY
Mr Long
Mobi: 0986883886 - 0905710588
Email: long.npb@ttlcorp.vn
Website: ttlcorp.vn
iBroadcast Pro is a complete end-to-end digital solution for professional video production and broadcasting. It includes modules for master control room operations, video ingest, media management, traffic scheduling, mobile broadcasting, IPTV distribution, and cinema playback. The software is built on SAO's SDK and uses the Infinity Core video server, which can seamlessly play multiple formats in the same playlist. iBroadcast Pro is designed to modernize broadcast workflows by replacing tape-based systems with an automated software solution.
This document summarizes a modular DSP resource board that complies with the AdvancedMC form factor standard. It can support up to 8 Texas Instruments C64xx DSPs and is designed for carrier grade telecom and other DSP-intensive applications. The board provides flexible and scalable hardware, integrated software, and a development environment to help customers integrate DSP capabilities and bring solutions to market quickly.
IPNext180 is a next generation hybrid IP-PBX system that provides:
- Powerful voice mail, IVR, and call handling features.
- Support for PSTN interfaces and various IP terminals.
- Fault tolerant and scalable architecture with system redundancy options.
- RTP proxy service for private IP networks, presence service for UC, and IVR script editor.
- Management of users, devices, calls, and networks through standard protocols.
The document describes the three user classes for the Cisco CME GUI:
1. System Administrator - Can configure all system-wide and phone features and is familiar with Cisco IOS software.
2. Customer Administrator - Can perform routine phone adds and changes without Cisco IOS software training.
3. Phone User - Can program a small set of phone features and search the directory.
CSS has established itself as the leader in Telephony & Call Recording solutions to the North American Security Industry.
We do not provide another plain old phone system, we do provide a solution designed specifically for the central station and interoffice communication, solutions which streamline operations, increase productivity and provide features no other product can provide.
The Avaya 1120E IP Deskphone is a four-line IP phone that supports VoIP and SIP protocols. It has a graphical display, integrated USB port, and Gigabit Ethernet for supporting multimedia applications. The phone offers presence, instant messaging and other collaborative features. It is compatible with Avaya and third-party communication servers and is suited for office workers needing robust communications capabilities.
Planning and Troubleshooting VoIP Performance shares insights on ThousandEyes helps visualize VoIP routing between branch offices and across the internet, optimize and plan new VoIP deployments and expansions, and troubleshoot VoIP performance to specific problem nodes, links and networks.
Developing Applications Using Host Processing Instead of DSPsVideoguy
Host media processing (HMP) involves using general-purpose computing platforms instead of specialized hardware to create telephony applications. This offers lower costs through reduced hardware requirements, more flexibility in application deployment, and increased reliability by distributing processing across multiple servers. HMP provides voice processing and media services through software running on standard servers.
SecurVoice Call Recording is the most advanced and powerful call recording solution. Interfaced to all the major automation software systems for event integration.
PLNOG14: Fortinet, Carrier and MSSP - Robert DąbrowskiPROIDEA
Robert Dąbrowski - Fortinet
Language: English
The presentation covers types of projects as well as specific examples of FORTINET activity in the telecommunications sector.
It showcases technologies, their development and advancement driven by the needs of service providers for securing the ISP infrastructure and MSSP service distribution.
Register to the next PLNOG edition today: krakow.plnog.pl
This document provides an overview of open source PBX software called Asterisk. It discusses VoIP technologies including codecs, protocols and PBX features. It also outlines how to install, configure and use Asterisk to set up a PBX system with channels, phones, IVRs and billing integration. Hardware requirements and options for interfaces are presented along with examples of configuration files. The document demonstrates how to register softphones and test calling between Asterisk and other VoIP systems.
The document provides technical specifications for BusinessClassSIPTrunks from Time Warner Cable. It outlines that an Enterprise SIP Gateway (ESG) is installed on the customer's premises and connected either directly to their IP PBX or to a switch. It lists requirements for the IP PBX configuration including IP addressing within a single subnet. It also specifies bandwidth, power, and physical space requirements for the ESG as well as protocols and codecs supported.
This document summarizes the current state of free and open source software (FOSS) in broadcast video applications. It notes that FOSS sees little use in broadcast due to large budgets and preference for proprietary solutions, though FOSS is widely used behind the scenes. It outlines some upsides like fitting into segmented broadcast workflows and convergence with IT. It highlights some current FOSS broadcast projects like CasparCG and Dirac. It then details the Open Broadcast Encoder project which aims to provide a free and open high-end broadcast video encoder as a free alternative to expensive proprietary encoders.
This document provides an overview and update on AudioCodes session border controller (SBC) products. It summarizes that AudioCodes is a market leader in SBCs, with the fastest growing market share. It highlights key SBC products like the Mediant 9000 and features such as advanced routing management, global partner strategy, and a comprehensive product portfolio. The document aims to showcase AudioCodes' SBC technology and momentum in the enterprise voice and data market.
The Polycom SoundPoint IP 670 is a desktop IP phone with a large color display, HD voice quality, and Gigabit Ethernet connectivity. It provides a rich visual interface for applications and productivity. Key features include an expandable 6-34 line capacity, support for USB and third party applications, and advanced call handling and network connectivity capabilities.
The document discusses CoreStor, an IP recording solution from Delma that can capture and record IP traffic, including VoIP packets. It describes various methods for capturing IP traffic, such as using span ports, port mirroring, conferencing, or custom gateways. CoreStor is designed to integrate seamlessly into existing systems and provide recording in a single chassis. It supports standard computer hardware and includes replay, administration, and analysis client software.
The document provides an overview of VoIP components, standards, architectures and implementation choices. It discusses key VoIP elements like terminals, packetized voice, soft switches, media servers, gateways, LANs/WANs and standards. It also describes common VoIP architectures for computer-computer, computer-phone, phone-phone communication over the internet. Finally, it outlines VoIP solutions for businesses using VoIP-enabled PBXs, replacing PBXs with softswitches, and hosted PBX solutions.
Audio codes solution for genesys sip contact centerLong Nguyen
CHUYÊN CUNG CẤP THIỆT BỊ VÀ GIẢI PHÁP VOIP
TIME TRUE LIFE TECHNOLOGY JOINT STOCK COMPANY
Mr Long
Mobi: 0986883886 - 0905710588
Email: long.npb@ttlcorp.vn
Website: ttlcorp.vn
iBroadcast Pro is a complete end-to-end digital solution for professional video production and broadcasting. It includes modules for master control room operations, video ingest, media management, traffic scheduling, mobile broadcasting, IPTV distribution, and cinema playback. The software is built on SAO's SDK and uses the Infinity Core video server, which can seamlessly play multiple formats in the same playlist. iBroadcast Pro is designed to modernize broadcast workflows by replacing tape-based systems with an automated software solution.
This document summarizes a modular DSP resource board that complies with the AdvancedMC form factor standard. It can support up to 8 Texas Instruments C64xx DSPs and is designed for carrier grade telecom and other DSP-intensive applications. The board provides flexible and scalable hardware, integrated software, and a development environment to help customers integrate DSP capabilities and bring solutions to market quickly.
IPNext180 is a next generation hybrid IP-PBX system that provides:
- Powerful voice mail, IVR, and call handling features.
- Support for PSTN interfaces and various IP terminals.
- Fault tolerant and scalable architecture with system redundancy options.
- RTP proxy service for private IP networks, presence service for UC, and IVR script editor.
- Management of users, devices, calls, and networks through standard protocols.
The document describes the three user classes for the Cisco CME GUI:
1. System Administrator - Can configure all system-wide and phone features and is familiar with Cisco IOS software.
2. Customer Administrator - Can perform routine phone adds and changes without Cisco IOS software training.
3. Phone User - Can program a small set of phone features and search the directory.
CSS has established itself as the leader in Telephony & Call Recording solutions to the North American Security Industry.
We do not provide another plain old phone system, we do provide a solution designed specifically for the central station and interoffice communication, solutions which streamline operations, increase productivity and provide features no other product can provide.
The Avaya 1120E IP Deskphone is a four-line IP phone that supports VoIP and SIP protocols. It has a graphical display, integrated USB port, and Gigabit Ethernet for supporting multimedia applications. The phone offers presence, instant messaging and other collaborative features. It is compatible with Avaya and third-party communication servers and is suited for office workers needing robust communications capabilities.
Planning and Troubleshooting VoIP Performance shares insights on ThousandEyes helps visualize VoIP routing between branch offices and across the internet, optimize and plan new VoIP deployments and expansions, and troubleshoot VoIP performance to specific problem nodes, links and networks.
Developing Applications Using Host Processing Instead of DSPsVideoguy
Host media processing (HMP) involves using general-purpose computing platforms instead of specialized hardware to create telephony applications. This offers lower costs through reduced hardware requirements, more flexibility in application deployment, and increased reliability by distributing processing across multiple servers. HMP provides voice processing and media services through software running on standard servers.
SecurVoice Call Recording is the most advanced and powerful call recording solution. Interfaced to all the major automation software systems for event integration.
PLNOG14: Fortinet, Carrier and MSSP - Robert DąbrowskiPROIDEA
Robert Dąbrowski - Fortinet
Language: English
The presentation covers types of projects as well as specific examples of FORTINET activity in the telecommunications sector.
It showcases technologies, their development and advancement driven by the needs of service providers for securing the ISP infrastructure and MSSP service distribution.
Register to the next PLNOG edition today: krakow.plnog.pl
This document provides an overview of open source PBX software called Asterisk. It discusses VoIP technologies including codecs, protocols and PBX features. It also outlines how to install, configure and use Asterisk to set up a PBX system with channels, phones, IVRs and billing integration. Hardware requirements and options for interfaces are presented along with examples of configuration files. The document demonstrates how to register softphones and test calling between Asterisk and other VoIP systems.
The document provides technical specifications for BusinessClassSIPTrunks from Time Warner Cable. It outlines that an Enterprise SIP Gateway (ESG) is installed on the customer's premises and connected either directly to their IP PBX or to a switch. It lists requirements for the IP PBX configuration including IP addressing within a single subnet. It also specifies bandwidth, power, and physical space requirements for the ESG as well as protocols and codecs supported.
This document summarizes the current state of free and open source software (FOSS) in broadcast video applications. It notes that FOSS sees little use in broadcast due to large budgets and preference for proprietary solutions, though FOSS is widely used behind the scenes. It outlines some upsides like fitting into segmented broadcast workflows and convergence with IT. It highlights some current FOSS broadcast projects like CasparCG and Dirac. It then details the Open Broadcast Encoder project which aims to provide a free and open high-end broadcast video encoder as a free alternative to expensive proprietary encoders.
This document provides an overview and update on AudioCodes session border controller (SBC) products. It summarizes that AudioCodes is a market leader in SBCs, with the fastest growing market share. It highlights key SBC products like the Mediant 9000 and features such as advanced routing management, global partner strategy, and a comprehensive product portfolio. The document aims to showcase AudioCodes' SBC technology and momentum in the enterprise voice and data market.
The Polycom SoundPoint IP 670 is a desktop IP phone with a large color display, HD voice quality, and Gigabit Ethernet connectivity. It provides a rich visual interface for applications and productivity. Key features include an expandable 6-34 line capacity, support for USB and third party applications, and advanced call handling and network connectivity capabilities.
7 reasons why video conferencing world will neverTrueConf
The video conferencing world will never be the same again due to several technological trends:
1. Advances in CPU technology allow for higher quality video encoding and decoding on regular computers and mobile devices.
2. Scalable video coding allows a single system to replace traditional MCU infrastructure, supporting multiple resolutions from SD to 4K without transcoding.
3. WebRTC enables real-time communications directly in web browsers without plugins.
4. Interoperability standards like SIP allow different video systems to connect with one another.
5. Consumerization trends like mobility, BYOD, and wireless interfaces are influencing enterprise video conferencing.
Advanced topologies for microsoft e learning shared by voip.com.vnTran Thanh
The document discusses advanced topology solutions for integrating AudioCodes media gateways with Microsoft unified communications networks and applications like OCS and Exchange. It provides examples of how AudioCodes gateways can enable connectivity between non-certified IP PBXs and Microsoft solutions, provide survivability in failure scenarios, and support migration to fully unified communications networks. The gateways provide flexibility, scalability, manageability and high availability.
The SoundPoint IP 601 is a VoIP phone that delivers high quality voice and advanced features. It supports up to 3 expansion modules for additional lines and features. The phone has an intuitive interface, dedicated feature keys, and a high resolution display. It provides productivity features like multiple lines, conferencing, and messaging. When equipped with expansion modules, it can function as a high-performance attendant console to manage many simultaneous calls.
1) Videoconferencing allows participants to see, hear and collaborate in real-time over networks or the internet. It requires equipment like cameras, microphones and displays.
2) Key standards like H.320, H.323 and SIP define how audio, video and data are transmitted over different networks like ISDN, IP and cellular. Codecs compress video and audio for efficient transmission.
3) Popular applications of videoconferencing include meetings, education, telemedicine and more. Proper etiquette like preparing agendas, camera positioning and avoiding distractions enhances collaboration.
- Videoconferencing allows participants to see, hear and collaborate in real time over internet or telephone networks. It requires equipment like cameras, microphones, displays and codecs to compress and decompress audio/video data.
- Standards like H.320, H.323 and H.324 specify protocols for videoconferencing over ISDN, IP networks and POTS lines. Transport methods include ISDN, IP networks, cellular networks and POTS lines.
- Key components of videoconferencing systems are video/audio input/output devices, data transfer networks, and codecs. Formats like H.261, H.263, H.264 and audio standards G.711, G.722 are
1) Videoconferencing allows participants to see, hear and collaborate in real time over networks or the internet. It requires equipment like cameras, microphones and displays.
2) Standards like H.320, H.323 and H.324 define protocols for videoconferencing over different mediums. Codecs compress audio and video for transmission. Transport protocols include TCP, UDP and RTP.
3) Popular applications of videoconferencing include meetings, education, telemedicine and more. Setup, quality and costs vary depending on the medium used such as ISDN, IP networks or cellular networks.
This document provides an overview of multimedia services over IP networks and discusses two key protocols used: SIP and H.323. It describes the basics of SIP including session descriptions using SDP, message format, and session initiation. It also discusses SIP applications like IMS including requirements, protocols used, and architecture. For H.323, it outlines the network architecture including terminals, MCUs, gateways, and gatekeepers. It then describes the H.323 signaling protocols including RAS, H.225 call signaling, and H.245 call control.
Surf Communication Solutions provides of MoP (Media over Packet) Triple Play (Voice, Video, and Modem/Fax/Data) conversion solutions to communication equipment manufacturers. These solutions are provided in various integration levels: DSP software ; PTMC boards; DSP hardware/software; and PCI boards. http://www.surf-com.com
What you really need to know about Video Conferencing SystemsVideoguy
This document discusses factors to consider when choosing a video conferencing system, including available bandwidth, acceptable quality levels, and supported standards. It outlines different connection types like ISDN, LAN/WAN, cellular networks and their associated standards. Newer standards like H.264 can provide better quality at lower bandwidths. The best system depends on expectations, bandwidth, number of participants, locations, management needs and costs.
This document discusses how VoIP systems work, including:
1) Soft switches translate phone numbers to IP addresses and know the current location and IP address of endpoints on the network.
2) Common VoIP protocols like H.323 and SIP are used to connect hardware and support real-time video, audio, and data applications. However, a lack of standardization can cause compatibility issues.
3) VoIP offers advantages over traditional phone systems like lower costs, mobility via internet connections, and reduced bandwidth requirements.
The Polycom SoundPoint IP 335 is an entry-level IP phone that provides excellent sound quality and enterprise-grade telephony features at an affordable price. It utilizes Polycom HD Voice technology and a high resolution display to deliver clear voice calls and an intuitive interface. The phone is designed for cubicle workers and call centers to enhance productivity through features such as conferencing and headset compatibility.
The document summarizes an IP phone system called Allworx that is designed for businesses. It provides diverse voice services, robust features, and global integration capabilities across multiple sites. Allworx systems aim to satisfy customer needs with low total cost of ownership through easy installation and maintenance.
Polycom soundstation ip6000 sip data sheetbest4systems
The Polycom SoundStation IP 6000 is an advanced IP conference phone designed for small to midsize conference rooms. It features Polycom HD Voice technology for crystal clear audio, a 12-foot microphone pickup area that can be expanded, and robust SIP compatibility. The phone provides superior audio quality, ease of use, and security for clear communication in conference calls.
The document discusses several voice and video platforms that use MCU/SFU concepts for real-time communication. It provides details on the Agora platform, Agora SDK, Radisys video conferencing system, Janus WebRTC gateway, SwitchRTC platform, Tokbox OpenTok platform, Twilio platform, and proposes an architecture to develop video conferencing. Key points include how these platforms support audio/video calls, use direct peer-to-peer or server-based connections, support different protocols and SDKs, and provide features like live streaming, video rooms, and transcription.
Teksun Microsys Pvt. Ltd. offers various product design and engineering services including automotive product development, mobile application development, hardware design, embedded software services, and application development. The company has experience in industries such as automotive, industrial systems, and consumer electronics. Services include electronics design, embedded software, application development, testing, and product deployment. Areas of expertise include microcontrollers, FPGAs, communication protocols, and operating systems such as Android and Linux. Example projects include an animal tracking system using RFID, a network video recorder, and an asset tracking system for workshops.
Surf Communication Solutions provides multimedia processing products and solutions that enable convergence of voice, video, and data across wired and wireless networks. The company was founded in 1996 and has over 60 employees worldwide focused on research and development. Surf's products include DSP boards, software, and chip-level solutions that support applications such as media gateways, media servers, and IP PBX systems.
Video Conferencing : Fundamentals and ApplicationVideoguy
The document discusses video conferencing fundamentals and applications. It covers topics like modes of video conferencing, components, technologies, standards, protocols, bandwidth requirements, quality of service factors, challenges, and the eBaithak desktop video conferencing system developed at IIT Kharagpur.
Generating privacy-protected synthetic data using Secludy and MilvusZilliz
During this demo, the founders of Secludy will demonstrate how their system utilizes Milvus to store and manipulate embeddings for generating privacy-protected synthetic data. Their approach not only maintains the confidentiality of the original data but also enhances the utility and scalability of LLMs under privacy constraints. Attendees, including machine learning engineers, data scientists, and data managers, will witness first-hand how Secludy's integration with Milvus empowers organizations to harness the power of LLMs securely and efficiently.
FREE A4 Cyber Security Awareness Posters-Social Engineering part 3Data Hops
Free A4 downloadable and printable Cyber Security, Social Engineering Safety and security Training Posters . Promote security awareness in the home or workplace. Lock them Out From training providers datahops.com
5th LF Energy Power Grid Model Meet-up SlidesDanBrown980551
5th Power Grid Model Meet-up
It is with great pleasure that we extend to you an invitation to the 5th Power Grid Model Meet-up, scheduled for 6th June 2024. This event will adopt a hybrid format, allowing participants to join us either through an online Mircosoft Teams session or in person at TU/e located at Den Dolech 2, Eindhoven, Netherlands. The meet-up will be hosted by Eindhoven University of Technology (TU/e), a research university specializing in engineering science & technology.
Power Grid Model
The global energy transition is placing new and unprecedented demands on Distribution System Operators (DSOs). Alongside upgrades to grid capacity, processes such as digitization, capacity optimization, and congestion management are becoming vital for delivering reliable services.
Power Grid Model is an open source project from Linux Foundation Energy and provides a calculation engine that is increasingly essential for DSOs. It offers a standards-based foundation enabling real-time power systems analysis, simulations of electrical power grids, and sophisticated what-if analysis. In addition, it enables in-depth studies and analysis of the electrical power grid’s behavior and performance. This comprehensive model incorporates essential factors such as power generation capacity, electrical losses, voltage levels, power flows, and system stability.
Power Grid Model is currently being applied in a wide variety of use cases, including grid planning, expansion, reliability, and congestion studies. It can also help in analyzing the impact of renewable energy integration, assessing the effects of disturbances or faults, and developing strategies for grid control and optimization.
What to expect
For the upcoming meetup we are organizing, we have an exciting lineup of activities planned:
-Insightful presentations covering two practical applications of the Power Grid Model.
-An update on the latest advancements in Power Grid -Model technology during the first and second quarters of 2024.
-An interactive brainstorming session to discuss and propose new feature requests.
-An opportunity to connect with fellow Power Grid Model enthusiasts and users.
Monitoring and Managing Anomaly Detection on OpenShift.pdfTosin Akinosho
Monitoring and Managing Anomaly Detection on OpenShift
Overview
Dive into the world of anomaly detection on edge devices with our comprehensive hands-on tutorial. This SlideShare presentation will guide you through the entire process, from data collection and model training to edge deployment and real-time monitoring. Perfect for those looking to implement robust anomaly detection systems on resource-constrained IoT/edge devices.
Key Topics Covered
1. Introduction to Anomaly Detection
- Understand the fundamentals of anomaly detection and its importance in identifying unusual behavior or failures in systems.
2. Understanding Edge (IoT)
- Learn about edge computing and IoT, and how they enable real-time data processing and decision-making at the source.
3. What is ArgoCD?
- Discover ArgoCD, a declarative, GitOps continuous delivery tool for Kubernetes, and its role in deploying applications on edge devices.
4. Deployment Using ArgoCD for Edge Devices
- Step-by-step guide on deploying anomaly detection models on edge devices using ArgoCD.
5. Introduction to Apache Kafka and S3
- Explore Apache Kafka for real-time data streaming and Amazon S3 for scalable storage solutions.
6. Viewing Kafka Messages in the Data Lake
- Learn how to view and analyze Kafka messages stored in a data lake for better insights.
7. What is Prometheus?
- Get to know Prometheus, an open-source monitoring and alerting toolkit, and its application in monitoring edge devices.
8. Monitoring Application Metrics with Prometheus
- Detailed instructions on setting up Prometheus to monitor the performance and health of your anomaly detection system.
9. What is Camel K?
- Introduction to Camel K, a lightweight integration framework built on Apache Camel, designed for Kubernetes.
10. Configuring Camel K Integrations for Data Pipelines
- Learn how to configure Camel K for seamless data pipeline integrations in your anomaly detection workflow.
11. What is a Jupyter Notebook?
- Overview of Jupyter Notebooks, an open-source web application for creating and sharing documents with live code, equations, visualizations, and narrative text.
12. Jupyter Notebooks with Code Examples
- Hands-on examples and code snippets in Jupyter Notebooks to help you implement and test anomaly detection models.
TrustArc Webinar - 2024 Global Privacy SurveyTrustArc
How does your privacy program stack up against your peers? What challenges are privacy teams tackling and prioritizing in 2024?
In the fifth annual Global Privacy Benchmarks Survey, we asked over 1,800 global privacy professionals and business executives to share their perspectives on the current state of privacy inside and outside of their organizations. This year’s report focused on emerging areas of importance for privacy and compliance professionals, including considerations and implications of Artificial Intelligence (AI) technologies, building brand trust, and different approaches for achieving higher privacy competence scores.
See how organizational priorities and strategic approaches to data security and privacy are evolving around the globe.
This webinar will review:
- The top 10 privacy insights from the fifth annual Global Privacy Benchmarks Survey
- The top challenges for privacy leaders, practitioners, and organizations in 2024
- Key themes to consider in developing and maintaining your privacy program
Digital Marketing Trends in 2024 | Guide for Staying AheadWask
https://www.wask.co/ebooks/digital-marketing-trends-in-2024
Feeling lost in the digital marketing whirlwind of 2024? Technology is changing, consumer habits are evolving, and staying ahead of the curve feels like a never-ending pursuit. This e-book is your compass. Dive into actionable insights to handle the complexities of modern marketing. From hyper-personalization to the power of user-generated content, learn how to build long-term relationships with your audience and unlock the secrets to success in the ever-shifting digital landscape.
Introduction of Cybersecurity with OSS at Code Europe 2024Hiroshi SHIBATA
I develop the Ruby programming language, RubyGems, and Bundler, which are package managers for Ruby. Today, I will introduce how to enhance the security of your application using open-source software (OSS) examples from Ruby and RubyGems.
The first topic is CVE (Common Vulnerabilities and Exposures). I have published CVEs many times. But what exactly is a CVE? I'll provide a basic understanding of CVEs and explain how to detect and handle vulnerabilities in OSS.
Next, let's discuss package managers. Package managers play a critical role in the OSS ecosystem. I'll explain how to manage library dependencies in your application.
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2. Contents
• Prosody S summary
• Why choose Prosody S?
– 6 reasons
• What can I build?
• Features summary
• Roadmap
3. The Prosody family of products
• DSP-based processing in board
and 1U box form-factors
• IP and TDM media
• VoIP and PSTN signalling
• Up to 16 E1/T1 trunks
• Software product
• IP media
• VoIP signalling
4. Prosody S summary
• Host media processing (HMP) software
• Runs on host CPU of standard IT servers
• Voice (VoIP) and fax communications applications
• Enterprise and service providers
• Scales from 1 to thousands of channels
5. Why choose Prosody S?
1. Very comprehensive media processing
capabilities
• Wide range of Codecs supported – including HD
Voice
• Extensive fax support T.30/T.38
• Deployment proven SIP stack
6. Why choose Prosody S?
2. Flexible deployment models
• Bare metal servers, virtual machines, hosted
servers or cloud servers
• A valid proposition to build up “ hosted “ ( Private
Cloud ) based SW solutions
• Linux & Windows, 32 & 64 bit
• Distributed architecture
• Flexible licensing – pay for what you need, per
channel
7. Why choose Prosody S?
3. Scalability
• Scales from 10s to thousands of channels
• Install only the channel quantity needed; support
for licence moves and changes to meet your
capacity needs
8. Why choose Prosody S?
4. Cost efficient
• Simple pricing, no hidden costs for extra
functionality
• Pay only for what you use (per channel)
• Supports startup deployments with low initial costs
• Capex or Opex purchase models (buy or rent)
9. Why choose Prosody S?
5. Robust, proven technology
• Trusted by organisations globally
• Thousands of deployed channels in both
enterprise and mission critical (emergency
services) scenarios
• Broad set of reference cases
10. Why choose Prosody S?
6. Simple transition for Prosody X customers
• Same API as Prosody X
• Same installation tools as Prosody X
11. • On-premise or hosted/cloud-based solutions
• SIP-based IP-PBX services
• Virtual PBXs
• IP contact centres
• Audio conferencing servers
• Fax servers
• Cloud communications services
What can I build?
12. • Voice API – play/record, DTMF handling, answering machine
detection, etc.
• Transcoding
• Conferencing
• Fax transport
• Wide range of fixed and mobile network audio codecs supported
including HD Voice codecs for optimal audio quality
• Works on Windows/Linux servers and virtual machines
• IPv4/IPv6 support
• Distributed (remote) API architecture
• Common API with Prosody X hardware
– For common features (e.g., no switch API)
• Dual redundant SIP service (DRSS)
Prosody S – key features summary
15. Prosody S – existing functionality
SIP/SDP
RTP
Signalling and
control:
Media layer:
G.711(I & II) G.723.1A G.726
MRCP
Secure RTP
G.728 G.729AB
GSM-FR GSM-EFR
T.38
H.323
Voice codecs:
Fax:
Recording Playback
Media
processing:
CNG
ConferencingTranscoding
PLC
VAD DTMF
handling
iLBC
AGC
Jitter buffer
AMR-NB
EVRC
Call progress
MS-GSM
OKI/IMA
ADPCM
Speex TETRA
RTCP
Live speaker
detection
Active speaker
notification
SIPS/TLS
G.722 G.722.1 G.722.2
(AMR-WB)
T.30 over G.711
SILK
16. Features explained – distributed API architecture
• Application software operates on centralised server
• Multiple Prosody S equipped servers supported, can
be scaled to meet implementation needs
– No impact to local compute resources on each Prosody S
server
• Flexible scaling of site capacity
– Operational cost control
– Quick response to expansion needs
• Operational integrity
– 3 levels of redundancy (plus DRSS)
– Failover for application control
17. Call centre application with distributed API
Software, distributed amongst several servers,
can act as one solution
Firewall
Public
IP network
Server #1
Prosody S
Local area network
Storage
database
Agent stations
Callers
Prosody S
Server #n
Control
application
Contact centre solution
18. Features explained – common API with Prosody X
• Developers have only one API set to learn
• Projects can be ported easily to Prosody X or
Prosody S as required
– If suitably programmed (no switch API)
– Professional Services available for consultancy advice
• Quicker feature development times for Aculab –
faster time to market for our customers
19. Features explained – fax
• Both G.711 fax pass through and T.38 fax modes are supported
• T.38 fax (FoIP) is intended to replace the sometimes un-reliable fax
over VoIP approach (G.711 pass through) and enable full migration
to an all-IP network for all business telecom needs
20. Features explained – wideband (HD Voice) support
• HD Voice codecs
– 20Hz to 7kHz audio bandwidth for clearer speech
applications
– G.722; G.722.1*; G.722.2 (AMR-WB) all available
– Skype SILK
– Others under consideration – EVRC-NW, Speex
wideband, Broadcom BV32, Opus
• Applications
– Wideband conferencing server
– Wideband enabled call centre server
– Transcoding server
– Gaming platforms
– etc
* G.722.1 is offered as a beta version awaiting full testing
21. Features explained – dual redundant SIP
service (DRSS)
• Ability to duplicate the SIP stacks for resilience
• Bringing PSTN centric ‘five 9s’ availability
concepts to the SIP world
– Fundamental requirement for telcos; becoming more important
now also for enterprises
• Recovery from failure conditions taken for
granted in the PSTN, but equally valid for SIP
networks
• Reactions from customers and analysts very
positive – we have something unique
• Whitepaper available via website
22. How to purchase
• Licence purchases will be credited to a customers licensing account
accessible 24x7
• Single channels or blocks of credits can be purchased in advance,
used as needed
• Licence keys created by the customer as required using a web
interface to our licensing database
• Rental option now available, contact us for details
• Prosody S software
is licensed, granular
increments
– Part code ACS9200
– Prosody S single
channel licence
23. Channel counts – Prosody S v3.0 (Windows)
Notes:
1. Typical figures as tested on Intel Core 2 Extreme (X6800) @ 2.93GHz running Windows XP SP3
2. All the channel count figures in the table are given for 85% CPU utilization
3. Refer to datasheet for full list of channel counts and codec types
4. Wideband codec tests performed on Intel Xeon quad-core machine
Prosody S
feature
Feature
detail
G.711
codec
G.723.1A
codec
G.729ab
codec
T.38 fax G.722
G.722.2
(AMR-
WB)
Music on hold
playback
10 different
music replays
2200 130 400
Recording and
playback
Full-duplex
channels
630 115 230 604 364
Recording and
playback
Full-duplex
channels with
DTMF
detection
580 115 220
Matrix
conferencing
Full-duplex
channels
730 120 250
Transcoding
to/from G.711
Full-duplex
channels
n/a 115 210
Fax (FoIP) 750
Windows OS
24. Channel counts – Prosody S v3.0 (Linux)
Notes:
1. Typical figures as tested on Intel Core 2 Extreme (X6800) @ 2.93GHz running SUSE Linux version 10.3
2. All the channel count figures in the table are given for 85% CPU utilization
3. Refer to datasheet for full list of channel counts and codec types
Prosody S
feature
Feature
detail
G.711
codec
G.723.1A
codec
G.729ab
codec
T.38
fax
G.722
G.722.2
(AMR-
WB)
Music on hold
playback
10 different music
replays
4000 130 650
Recording and
playback
Full-duplex
channels
850 120 310 110 40
Recording and
playback
Full-duplex
channels with
DTMF detection
750 120 310
Matrix
conferencing
Full-duplex
channels
800 120 360
Transcoding
to/from G.711
codec
Full-duplex
channels
n/a 120 340
Fax (FoIP) 1600
Linux OS