The document introduces the TesiraFORTÉ family of audio DSP devices from Biamp. It describes the four models - AI, CI, TI, and VI - which differ in their integrated audio conferencing capabilities like acoustic echo cancellation and support for VoIP or standard telephone lines. The models all provide audio processing and networking functionality through a fixed I/O configuration in a 1RU form factor. The document provides details on the features and intended use cases of each TesiraFORTÉ model.
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.
This document provides an overview of streaming video over IP networks for commercial audiovisual applications. It discusses video over IP, hardware and software options, common codecs and streaming protocols, and network considerations. It also covers integration with control systems and the benefits of using a standard Ethernet protocol like JPEG2000 for applications requiring low latency video distribution over a local area network.
Keyboard, Video And Mouse (KVM) Switch SolutionPremier Farnell
The document discusses KVM switch technologies that allow multiple computers to be accessed from a single keyboard, mouse and monitor. It analyzes different types of KVM switches including analog and digital and recommends components like video buffers, amplifiers, switches and processors that can be used to build a KVM switch solution. Key aspects covered include types of KVM switches, applications, technologies, component selection guides and solutions.
The document summarizes new products and features being displayed by AETA Audio Systems at the 2012 IBC Show, including:
1) The ScoopFone HD, a professional mobile phone that integrates phone calls into the broadcast chain and offers improved audio quality of up to 7kHz.
2) The Scoop5 codec, the newest generation of Scoop technology with upgraded software and ergonomic design.
3) The 4MinX integrated digital mixer and recorder, which won an innovation award, now has improved usability.
Polycom sound station ip4000 data sheetbest4systems
The Polycom SoundStation IP 4000 SIP is a conference phone designed for SIP environments that provides improved audio quality over previous models. It allows natural conversations from up to 10 feet away and includes features like optional extended microphones to expand coverage, gated microphones to reduce echo and noise, and a backlit display for easy menu navigation. The phone provides familiar call functions and connects directly into SIP systems for high-quality voice conferencing.
Polycom soundpoint ip670 add on module data sheetbest4systems
The SoundPoint IP Color Expansion Module enhances productivity for telephone attendants by providing a large color display, 14 programmable line keys, and advanced call handling capabilities. It easily attaches to the SoundPoint IP 670 phone without additional setup needed. Up to three Expansion Modules can be used with a single phone to create a powerful attendant console.
This document discusses planning for video streaming over IT networks. It describes treating video streams as just another type of data packet. It provides bandwidth requirements for uncompressed and compressed video formats. It discusses SVSI's N-Series encoders/decoders that use various compression formats suitable for different network applications. It also provides examples of N-Series installations in corporate campuses, casinos, universities and more.
This document summarizes information presented by AudioCodes about migrating to SIP trunking. It discusses:
1) How AudioCodes solutions like media gateways and session border controllers enable businesses to migrate phone systems to SIP trunking gradually, starting with a trial that reduces reliance on PSTN lines.
2) The cost savings that SIP trunking can provide compared to traditional phone lines, often 25-50% lower recurring charges through elimination of dual infrastructure and competitive SIP trunking markets.
3) A promotional offer from AudioCodes for VARs, which provides discounted hardware and support, as well as SIP training and certification, to help customers evaluate SIP trunking.
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.
This document provides an overview of streaming video over IP networks for commercial audiovisual applications. It discusses video over IP, hardware and software options, common codecs and streaming protocols, and network considerations. It also covers integration with control systems and the benefits of using a standard Ethernet protocol like JPEG2000 for applications requiring low latency video distribution over a local area network.
Keyboard, Video And Mouse (KVM) Switch SolutionPremier Farnell
The document discusses KVM switch technologies that allow multiple computers to be accessed from a single keyboard, mouse and monitor. It analyzes different types of KVM switches including analog and digital and recommends components like video buffers, amplifiers, switches and processors that can be used to build a KVM switch solution. Key aspects covered include types of KVM switches, applications, technologies, component selection guides and solutions.
The document summarizes new products and features being displayed by AETA Audio Systems at the 2012 IBC Show, including:
1) The ScoopFone HD, a professional mobile phone that integrates phone calls into the broadcast chain and offers improved audio quality of up to 7kHz.
2) The Scoop5 codec, the newest generation of Scoop technology with upgraded software and ergonomic design.
3) The 4MinX integrated digital mixer and recorder, which won an innovation award, now has improved usability.
Polycom sound station ip4000 data sheetbest4systems
The Polycom SoundStation IP 4000 SIP is a conference phone designed for SIP environments that provides improved audio quality over previous models. It allows natural conversations from up to 10 feet away and includes features like optional extended microphones to expand coverage, gated microphones to reduce echo and noise, and a backlit display for easy menu navigation. The phone provides familiar call functions and connects directly into SIP systems for high-quality voice conferencing.
Polycom soundpoint ip670 add on module data sheetbest4systems
The SoundPoint IP Color Expansion Module enhances productivity for telephone attendants by providing a large color display, 14 programmable line keys, and advanced call handling capabilities. It easily attaches to the SoundPoint IP 670 phone without additional setup needed. Up to three Expansion Modules can be used with a single phone to create a powerful attendant console.
This document discusses planning for video streaming over IT networks. It describes treating video streams as just another type of data packet. It provides bandwidth requirements for uncompressed and compressed video formats. It discusses SVSI's N-Series encoders/decoders that use various compression formats suitable for different network applications. It also provides examples of N-Series installations in corporate campuses, casinos, universities and more.
This document summarizes information presented by AudioCodes about migrating to SIP trunking. It discusses:
1) How AudioCodes solutions like media gateways and session border controllers enable businesses to migrate phone systems to SIP trunking gradually, starting with a trial that reduces reliance on PSTN lines.
2) The cost savings that SIP trunking can provide compared to traditional phone lines, often 25-50% lower recurring charges through elimination of dual infrastructure and competitive SIP trunking markets.
3) A promotional offer from AudioCodes for VARs, which provides discounted hardware and support, as well as SIP training and certification, to help customers evaluate SIP trunking.
The document summarizes the key features and benefits of the Panasonic KX-NS500 Smart Hybrid PBX system. The system provides flexibility and scalability through a single, unified solution that allows businesses to minimize upfront costs, have a future-proof system, and consolidate multiple functions into one package. It offers cost savings through using existing equipment and lines, supports future business needs through expandability and mobility features, and ensures a simplified maintenance process.
MyPBX is a standalone embedded hybrid PBX for small businesses and remote branch offices that offers both VoIP and traditional phone system solutions. It supports up to 100 users with 22 concurrent calls and has features such as voicemail, conferencing, call forwarding, and call recording. The compact device has Ethernet, analog, and E1/T1 ports and supports codecs like G.711, G.722, and G.729.
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.
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 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 Yealink VP2009 Videophone is an IP multimedia communication device equipped with a 7-inch LCD display, 300K pixel CMOS camera, and TI DaVinci chipset. It supports H.264 video codecs and adaptive bandwidth adjustment to provide excellent video call quality even at low bandwidths. Additional features include a USB port, SD card slot, and external interfaces to enable functions like online/offline advertising.
The document discusses AudioCodes' range of hybrid E-SBC and media gateway products for connecting unified communications solutions to PBXs and legacy systems. It describes various models that support between 100 and 16,000 sessions and can be deployed in small branch offices or large enterprises. The products offer flexibility, scalability, and high availability options.
MyPBX E1 is a standalone embedded hybrid PBX for small businesses and remote branch offices that offers both VoIP and traditional telephony solutions. It supports up to 100 users and 22 concurrent calls, has E1/T1/J1 ports, 8 analog ports, and features like voicemail, conferencing, and call routing. The compact device has ample processing power and memory as well as standard and wideband codecs for high quality calls.
MyPBX is a standalone embedded hybrid PBX for small businesses and remote branch offices that offers a combination of VoIP and legacy telecom equipment. It provides features such as auto provisioning, call forwarding, conferencing, voicemail, and a web-based interface. The document provides specifications and contact information for obtaining demonstrations or further information.
The document discusses the open source Asterisk PBX software. It provides an overview of Asterisk including that it was created in 1999 as a free and open source alternative to expensive proprietary PBX systems. Asterisk allows users to build their own software-based phone systems using inexpensive hardware and can provide many of the same features as traditional PBXs through its flexible architecture and extensive capabilities. The document outlines some of Asterisk's main functionalities and how it works as well as hardware that can be used with it.
Its versatile design makes it ideal for multiple applications including videoconferencing, professional audio conferencing, wireless presentations and collaborations, training and corporate announcements etc.
COLLABORATE Pro 900 is perfect for both single-site collaboration and multi-site collaboration with up to 25 locations where users can join from anywhere, using any device – pc, mobile, room endpoints, or telephone.
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.
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.
Echo occurs when sound from speakers is picked up by microphones, such as from input and output audio devices placed close together. Various solutions exist to address echo issues in WebRTC implementations, including ensuring distance between devices, using headphones instead of built-in speakers, installing echo cancellation software, and adjusting audio constraints and volume controls. However, echo remains a challenging problem, particularly on mobile devices, as acoustic echo cancellation may not work or provide good quality for all use cases and device configurations. Ongoing work continues toward improving echo cancellation techniques in browsers.
Technoserve is a leading provider of IPTV and digital signage systems in Qatar and the Middle East. It has been in business for 14 years and has numerous clients. The document provides information on Technoserve's major clients, services, and products. It describes IPTV, including what it is, sample diagrams, features and products. It also describes digital signage in similar detail. The document concludes with information on Technoserve's support and service level agreement options.
This document introduces Denwa Product's unified communication platform and product lines. It provides an overview of Denwa's IP-PBX solutions for small and medium businesses, as well as larger corporate customers. The solutions include IP phones, video conferencing equipment, wireless phones, and desktop software for unified communications. Key features highlighted are HD voice, security capabilities like VPN, and integration with other collaboration tools.
http://www.televic-conference.com/en/unicos_multimedia_conference_system
uniCOS is a multimedia conference system that delivers interactivity and low-latency HD video to the conference delegates via a touch screen interface. The system has a dedicated, high-resilience network and communicates to the outside world via open standards. The Dante (TM) interface enables the uniCOS system to be integrated in a digital audio network.
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
H8118 HD HDMI encoder is a video and audio encoding product that supports:
- 1 HDMI video input channel and 1 independent audio input
- H.265 and H.264 video encoding and MP3, AAC, G711 audio formats
- Encoding and streaming video at resolutions up to 1920x1080p over networks using protocols like RTSP, RTMP, and ONVIF
The Octopus is an audio conferencing mixer that uses beamforming technology to deliver clear audio. It has four microphone inputs that can also be used for auxiliary devices or sound reinforcement. The Octopus allows for flexible microphone and speaker placement and can be daisy chained for more inputs. It connects via USB, RCA, or optional PSTN or DTI interfaces, and can include a built-in amplifier. The document provides an overview of the Octopus and its accessories.
HARMAN-SolutionGuide-Education-Conference Rooms Final 6915Mark Josiah Henkin
The document provides information on audio visual system solutions for conference rooms, including descriptions of common conference room activities and the core components of AV control & automation systems, sound reinforcement systems, collaboration & conferencing systems, and room scheduling & AV asset management systems. It also includes tips and concepts for designing integrated AV solutions for conference rooms using Harman's portfolio of professional audio and video technologies.
The document summarizes the key features and benefits of the Panasonic KX-NS500 Smart Hybrid PBX system. The system provides flexibility and scalability through a single, unified solution that allows businesses to minimize upfront costs, have a future-proof system, and consolidate multiple functions into one package. It offers cost savings through using existing equipment and lines, supports future business needs through expandability and mobility features, and ensures a simplified maintenance process.
MyPBX is a standalone embedded hybrid PBX for small businesses and remote branch offices that offers both VoIP and traditional phone system solutions. It supports up to 100 users with 22 concurrent calls and has features such as voicemail, conferencing, call forwarding, and call recording. The compact device has Ethernet, analog, and E1/T1 ports and supports codecs like G.711, G.722, and G.729.
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.
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 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 Yealink VP2009 Videophone is an IP multimedia communication device equipped with a 7-inch LCD display, 300K pixel CMOS camera, and TI DaVinci chipset. It supports H.264 video codecs and adaptive bandwidth adjustment to provide excellent video call quality even at low bandwidths. Additional features include a USB port, SD card slot, and external interfaces to enable functions like online/offline advertising.
The document discusses AudioCodes' range of hybrid E-SBC and media gateway products for connecting unified communications solutions to PBXs and legacy systems. It describes various models that support between 100 and 16,000 sessions and can be deployed in small branch offices or large enterprises. The products offer flexibility, scalability, and high availability options.
MyPBX E1 is a standalone embedded hybrid PBX for small businesses and remote branch offices that offers both VoIP and traditional telephony solutions. It supports up to 100 users and 22 concurrent calls, has E1/T1/J1 ports, 8 analog ports, and features like voicemail, conferencing, and call routing. The compact device has ample processing power and memory as well as standard and wideband codecs for high quality calls.
MyPBX is a standalone embedded hybrid PBX for small businesses and remote branch offices that offers a combination of VoIP and legacy telecom equipment. It provides features such as auto provisioning, call forwarding, conferencing, voicemail, and a web-based interface. The document provides specifications and contact information for obtaining demonstrations or further information.
The document discusses the open source Asterisk PBX software. It provides an overview of Asterisk including that it was created in 1999 as a free and open source alternative to expensive proprietary PBX systems. Asterisk allows users to build their own software-based phone systems using inexpensive hardware and can provide many of the same features as traditional PBXs through its flexible architecture and extensive capabilities. The document outlines some of Asterisk's main functionalities and how it works as well as hardware that can be used with it.
Its versatile design makes it ideal for multiple applications including videoconferencing, professional audio conferencing, wireless presentations and collaborations, training and corporate announcements etc.
COLLABORATE Pro 900 is perfect for both single-site collaboration and multi-site collaboration with up to 25 locations where users can join from anywhere, using any device – pc, mobile, room endpoints, or telephone.
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.
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.
Echo occurs when sound from speakers is picked up by microphones, such as from input and output audio devices placed close together. Various solutions exist to address echo issues in WebRTC implementations, including ensuring distance between devices, using headphones instead of built-in speakers, installing echo cancellation software, and adjusting audio constraints and volume controls. However, echo remains a challenging problem, particularly on mobile devices, as acoustic echo cancellation may not work or provide good quality for all use cases and device configurations. Ongoing work continues toward improving echo cancellation techniques in browsers.
Technoserve is a leading provider of IPTV and digital signage systems in Qatar and the Middle East. It has been in business for 14 years and has numerous clients. The document provides information on Technoserve's major clients, services, and products. It describes IPTV, including what it is, sample diagrams, features and products. It also describes digital signage in similar detail. The document concludes with information on Technoserve's support and service level agreement options.
This document introduces Denwa Product's unified communication platform and product lines. It provides an overview of Denwa's IP-PBX solutions for small and medium businesses, as well as larger corporate customers. The solutions include IP phones, video conferencing equipment, wireless phones, and desktop software for unified communications. Key features highlighted are HD voice, security capabilities like VPN, and integration with other collaboration tools.
http://www.televic-conference.com/en/unicos_multimedia_conference_system
uniCOS is a multimedia conference system that delivers interactivity and low-latency HD video to the conference delegates via a touch screen interface. The system has a dedicated, high-resilience network and communicates to the outside world via open standards. The Dante (TM) interface enables the uniCOS system to be integrated in a digital audio network.
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
H8118 HD HDMI encoder is a video and audio encoding product that supports:
- 1 HDMI video input channel and 1 independent audio input
- H.265 and H.264 video encoding and MP3, AAC, G711 audio formats
- Encoding and streaming video at resolutions up to 1920x1080p over networks using protocols like RTSP, RTMP, and ONVIF
The Octopus is an audio conferencing mixer that uses beamforming technology to deliver clear audio. It has four microphone inputs that can also be used for auxiliary devices or sound reinforcement. The Octopus allows for flexible microphone and speaker placement and can be daisy chained for more inputs. It connects via USB, RCA, or optional PSTN or DTI interfaces, and can include a built-in amplifier. The document provides an overview of the Octopus and its accessories.
HARMAN-SolutionGuide-Education-Conference Rooms Final 6915Mark Josiah Henkin
The document provides information on audio visual system solutions for conference rooms, including descriptions of common conference room activities and the core components of AV control & automation systems, sound reinforcement systems, collaboration & conferencing systems, and room scheduling & AV asset management systems. It also includes tips and concepts for designing integrated AV solutions for conference rooms using Harman's portfolio of professional audio and video technologies.
This document provides a summary of Biamp's DSP-functional product family. It discusses their everyday and enterprise audio solutions, including their basic DSP platforms like Nexia and Audia. It describes how they provide customizable solutions to fit customers' exact needs. It also discusses their flagship DSP platform, Tesira, and how it offers highly scalable and flexible networked audio solutions using AVB networking. The document includes examples of installations and provides a timeline of Biamp product releases.
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.
Hexolabs provides customizable IVR solutions using their takkit platform to reduce call costs for companies. They have expertise developing successful voice applications and solutions using their HEX 101 server with FreeSwitch software. The HEX 101 supports E1/T1/J1 telephony interfaces and Intel Core i7 architecture to deliver high performance for IP PBX, IVR, voice, and call center applications.
Embedded voice logger generally is called as non PC based voice recording system.Aria Telecom offers embedded voice logger in Delhi NCR. The system works as standalone tool and does not require computer to record voice conversation.
This document summarizes and describes the product portfolio of Vocality, a company that provides communications solutions. It outlines their Pro Series, BASICS Series, and OPUS Suite product lines. The Pro Series includes multiplexer and router devices in various form factors that provide voice, data, and network connectivity. The BASICS Series offers smaller and more portable devices that also provide voice, data, and network connectivity across different technologies. The OPUS Suite is a set of software products that provide network acceleration, optimization, security and multi-bearer functionality.
The SpeakerLinX SL254 is a decentralized IP audio amplifier and room controller that allows for high quality distributed audio systems. It contains hardware to convert IP audio signals to analog and amplify zones, while also controlling peripheral devices in each zone. By placing the amplifier close to speakers, the SL254 minimizes signal loss for better sound quality. It supports various audio formats and controls systems through a web interface. The SL254 provides 25 watts of power per channel to IP-ready speakers for each zone.
The IDX 200 is a fully integrated, networked audio communication system from HARMAN Professional that can distribute prerecorded messages, live paging, and audio content to up to 256 zones. It uses an open architecture software platform combined with DSP hardware and can integrate with amplifiers, speakers, microphones and other audio equipment. The scalable system allows expansion of inputs, DSP capabilities, zones and paging stations as needs grow.
The FRAFOS ABC Session Border Controller combines secure border control, signaling mediation, call routing, and advanced media server applications. It provides a customizable and scalable solution for securing peering connections and subscriber access for VoIP and NGN operators. As an open platform, the ABC SBC can be adapted according to customer needs and deployed on various hardware sizes to accommodate performance requirements of operators of all sizes.
Has video really killed the audio star?Cisco Canada
Video has drastically transformed the way we work with remote teams, business partners and customers. We have gone from faceless “who just joined?” audio only solutions to HD quality “better than being there” video options that foster active participation no matter your location or device.
Cisco has modernized and simplified our Video solutions. In this session, we will cover our repertoire of end points (from the pocket to the boardroom), the infrastructure that powers these end points (cloud, hybrid and on-premise), and the integration with other collaboration tools and applications (interoperability with Cisco and other vendor soft phones, hard phones, and conferencing).
Four types of Dante AVIO Adapters are available, including one- and two-channel versions of the analog input and analog output adapters.
More details visit: http://www.tm-systems.com.au/products/dante-avio-adaptors/
This document describes Media Networking Technology's Dante AVIO adapters which allow various audio equipment to be connected to a Dante audio network. The Dante AVIO adapters come in analog input/output, AES3, and USB versions to interface legacy audio gear with a Dante network. The adapters are plug-and-play, rugged for touring use, and provide high quality audio streaming and synchronization over the Dante network standard.
The document provides an overview of Bosch's portfolio of conference solutions, including the DICENTIS, DICENTIS Wireless, DCN, and CCS 1000 D systems. It highlights key features such as simultaneous interpretation for up to 100 languages, IP-based and wireless solutions, flexible installation options including flush mounting, integrated control systems, and audio recording capabilities. The document emphasizes Bosch's experience developing state-of-the-art and secure conference solutions to meet a wide range of customer needs from small local events to large international conferences.
The document introduces several new audio products for 2019 including the NMP40 network audio player module, AUDAC Touch control platform version 2.0, AMP203 mini stereo amplifier, ARU2xx relay modules, and CENA and CIRA ceiling speaker series. Key features highlighted are streaming audio directly to sound systems, customizable control interfaces, Dante digital audio input, PoE support, and fast installation systems for ceiling speakers.
Architectural media systems_application_guide_-_sporting_venuesDaniel Andozia
This document provides a summary of the audio system setup at a sporting venue. A Dante-enabled Soundcraft Vi1 console routes audio over Ethernet to multiple BSS Soundweb London devices located in equipment racks around the venue. The Soundweb devices process zone audio and send it over BLU Link to Crown DriveCore amplifiers, which power speakers in 16 zones covering seating areas as well as utility spaces. Wireless microphones connect directly to the console and Soundweb devices to provide announcements. The HiQnet Audio Architect interface allows remote control and monitoring of the system.
Architectural media systems_application_guide_-_sporting_venuesDaniel Andozia
This document provides a summary of the audio system setup at a sporting venue. A Dante-enabled Soundcraft Vi1 console routes audio over Ethernet to multiple BSS Soundweb London devices located in equipment racks around the venue. The Soundweb devices process zone audio and send it over BLU Link to Crown DriveCore amplifiers, which power speakers in 16 zones covering seating areas as well as utility spaces. Wireless microphones connect directly to the console and Soundweb devices to provide announcements. The HiQnet Audio Architect software allows remote control and monitoring of the entire system.
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.
*astTECS provides an open source video conferencing solution for virtual meetings. Key features include unlimited meetings per month, screen sharing, recording and playback of sessions, and integrations for file sharing and collaboration. It offers customizable solutions for small businesses and enterprises on affordable pricing plans.
Similar to Biamp_Product_Brochure_TesiraFORTE (20)
2. YOUR FORTÉ IS INSTALLING AUDIO SYSTEMS.
OURS IS MAKING YOUR AUDIO SOUND GREAT. NO MATTER WHAT.
When we created Tesira®
, it was always part of our plan to complement
our enterprise-level platform with devices built for everyday audio. The
TesiraFORTÉ family is the next step in our plan: four different models
with optional AVB networking, and optimized for specific applications
with fixed I/O configurations.
By combining our experience with market feedback, we’ve identified
a need for this kind of audio solution from our user community. Each
TesiraFORTÉ model is designed and optimized to address a different
business challenge, with an eye towards accommodating evolutions in
audio technologies.
We also recognize that some applications don’t require audio
networking. That’s why we offer an AVB and a non-AVB version for each
model. Don’t worry, both versions offer the same full set of features. The
only difference is the ability to transport audio using AVB.
NOT EVERY PROJECT REQUIRES NETWORKED AUDIO,
BUT EVERY PROJECT REQUIRES GREAT SOUND. [[
01 |
3. AVB PREPARES YOU FOR TOMORROW
AND TOMORROW STARTED YESTERDAY.
Audio Video Bridging (AVB) is a collection of IEEE 802.1 open standard
networking protocols that allow media streams to be carried over
Ethernet networks (using existing cable infrastructure) alongside
traditional data without interference. Networking with AVB leverages the
intelligence of Ethernet switches to the point where the switch actually
becomes an active, intuitive part of the AV system. It automatically
handles all the QoS, bandwidth reservation, mixed media usage, and
operates with a very low latency.
The wider adoption of AVB is a networking trend that continues to grow.
With the full support of the AVnu Alliance and innovative companies like
Biamp®
, Intel®
, Bose®
, Cisco®
, and Broadcom®
, AVB is the most advanced
media data management and processing technology available today.
By choosing an AVB-enabled
processor, you’re choosing a
smarter, more efficient audio
solution for your customers,
and one that will last them
long into the future.
| 02
4. To enhance the functionality and flexibility of TesiraFORTÉ, we added a USB port to
each model. The port can interface directly with third-party technologies, allowing you
to take full advantage of today’s most sophisticated conferencing solutions. Here are a
few applications where you may want to consider using USB audio:
THE FIRST AVB DSP WITH USB AUDIO.
CONFERENCING | Connect directly with a soft codec device
COURT RECORDING | Interface directly with court recording devices
03 |
1 AUDIO CHANNEL OUT OF TESIRAFORTÉ
1 AUDIO CHANNEL INTO TESIRAFORTÉ
SOFT CODEC | LAPTOP 01
SOFT CODEC | DESKTOP 02
MICS SPEAKERS
USB AUDIO
4, 6 or 8 AUDIO CHANNELS
OUT OF TESIRAFORTÉ
HARD DISC RECORDING SYSTEM 01
COMPUTER-BASED RECORDING SOFTWARE 02
COURT RECORDING SYSTEM 03
MICS SPEAKERS
USB AUDIO
5. CUTTING EDGE, ANYONE?
Each TesiraFORTÉ model supports up to 8 channels of configurable audio I/O that
can be allocated in the programming interface. With multiple combinations of inputs
and outputs, transporting audio over USB makes TesiraFORTÉ a powerful solution for
a number of common business challenges.
BACKGROUND MUSIC | Receive digital audio directly from a music source
LECTURE/MEETING RECORDING | Interface directly with recording devices
| 04
ADDITIONAL
AUDIO INPUTS
SPEAKERS
2 AUDIO CHANNELS
INTO TESIRAFORTÉ
MUSIC SERVICE PLAYER 01
COMPUTER WITH PLAYLIST 02
USB AUDIO
2 - 8 AUDIO CHANNELS
OUT OF TESIRAFORTÉ
HARD DISC RECORDING SYSTEM 01
COMPUTER-BASED RECORDING SOFTWARE 02
LECTURE CAPTURE SYSTEM 03
SPEAKERSMICS
USB AUDIO
6. • 128 x 128 channels of AVB
• 12 mic/line level inputs,
8 mic/line level outputs
• Gigabit Ethernet port
• Up to 8 channels of configurable
USB audio
• RS-232 serial port
TesiraFORTÉ AVB CI: External Codec Conferencing Solution
• Includes all features of the standard AI model, plus Sona™ AEC technology on all
12 inputs
TesiraFORTÉ AVB TI: Conferencing over Standard Telephone Service Solution
• Includes all features of the standard AI model, plus Sona AEC technology on all 12
inputs and standard telephone interface via an RJ-11 input connector
TesiraFORTÉ AVB VI: VoIP Conferencing Solution
• Includes all features of the standard AI model, plus Sona AEC technology on all 12
inputs and VoIP connectivity via an RJ-45 input connector
WHY ARE THERE
SO MANY TESIRAFORTÉS?
Because you shouldn't have to pay for
functionality you don’t need.
That’s the answer in a nutshell. Some audio DSP platforms perform
best in small, non-networked spaces, while others do better in
distributed, networked installations. TesiraFORTÉ lets you decide
what’s right for your project and budget.
Here are the specific features of each model:
TesiraFORTÉ AVB AI: Standard Model for High-Quality Audio
• 4-pin GPIO
• 2-line OLED display with capacitive-
touch navigation
• Rack mountable (1RU)
• Internal universal power supply
• Tesira server-class device
05 |
7. The non-AVB models have all the same functionalities listed previously, except those
specifically related to AVB networking. They can still interface with third-party controls and
be controlled and programmed remotely over a network.
The non-AVB TesiraFORTÉ models offer a cost-competitive, quality audio solution for
environments where audio networking isn’t needed. These models are ideal for conference
rooms that require their own audio DSP in the room.
TesiraFORTÉ hosts a Tesira configuration file and, therefore, does not
require a SERVER or SERVER-IO in order to function. TesiraFORTÉ can
also host a Biamp Canvas™ file and can be controlled by Biamp
Canvas interfaces. The AVB models can host all expander-
class devices (audio and logic expanders) as well as
send firmware updates via the Ethernet port. All
TesiraFORTÉ models can host Tesira TEC-1 control
devices, EX-LOGIC devices and can connect
via Ethernet or RS-232 with third-party control
systems. TesiraFORTÉ is accessible locally or
remotely over Ethernet for management
and maintenance.
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TESIRAFORTÉ
IS A SERVER-
CLASS DEVICE BY
DEFINITION
8. TesiraFORTÉ AVB AI
STANDARD MODEL FOR HIGH-QUALITY AUDIO
At its best in small- to medium-sized rooms such as conference rooms or council chambers,
the sweet spot for the AI model is installations requiring high-quality audio solutions such
as those with voice lift and mix-minus.
For those customers who want to link multiple rooms together for centralized control, or
add them to an existing Tesira AVB network, TesiraFORTÉ AVB AI models are the right
choice for general audio applications.
The TesiraFORTÉ AI is a better choice for those environments where a standalone DSP is
needed for localized audio per room.
TesiraFORTÉ AVB CI
EXTERNAL CODEC CONFERENCING SOLUTION
Soft codec conferencing is quickly becoming the way the world does business, and
TesiraFORTÉ CI is the ideal model to help you get the most from this technology. By
passing all audio signals through a powerful DSP that’s built to make sure every word is
heard, you get the best of both worlds.
With the USB port enabling direct integration with soft codec technologies, and AEC
technology ensuring superior conferencing audio, both the AVB and non-AVB CI models are
built for truly collaborative communications.
07 |
11. TesiraFORTÉ AVB TI
CONFERENCING OVER STANDARD TELEPHONE SERVICE SOLUTION
The TesiraFORTÉ TI models are built to provide specialized conferencing capabilities using
standard telephone service. For those customers who conference over analog telephone
lines, the TI model is a cost-effective solution for getting quality audio in their conference
rooms, without having to pay the cost of reconstructing existing infrastructure.
Whether you plan to link multiple rooms together, add them to an existing Tesira AVB
network, or keep the units separate, the TI models facilitate broad communication over
standard telephone lines and leverage AEC, with or without AVB.
TesiraFORTÉ AVB VI
VOIP CONFERENCING SOLUTION
Equipped with AEC and VoIP technologies, the TesiraFORTÉ VI models were born for
conferencing. Audio solutions using VoIP can be challenging, complex, and time-consuming
to design and install. With the VI, VoIP teleconferencing has never been easier.
We’ve leveraged our years of experience creating high-quality VoIP solutions to bring
you the VI model for high-level VoIP teleconferencing. Having the USB port and VoIP
capabilities gives you two options for conferencing, and even more ways to manage
your network.
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12. SO MANY TESIRAFORTÉS TO CHOOSE FROM.
AEC
VOIP CONFERENCING
STANDARD TELEPHONE CONFERENCING
AUDIO NETWORKING (AVB MODELS ONLY)
A
I
CI
TI
VI
FUNCTIONALITY
All
MODELS
ARE
EXCELLENT
FOR:
GENERAL ROOM AUDIO
BACKGROUND MUSIC
COURT RECORDING
LECTURE/MEETING RECORDING
VOICE LIFT AND/OR MIX-MINUS
SOFT CODEC INTEGRATION
All of our models are well-suited for many applications, from background music to soft
codec integration. The chart below illustrates some of the differentiating factors for
each model.
11 |
13. AVB MODELS
TesiraFORTÉ AVB AI: A digital server-class device
with fixed I/O configuration of 12 mic/line level
inputs and 8 mic/line level outputs, 4 GPIO
connections, up to 8 channels of configurable USB
audio, and networked audio via AVB.
TesiraFORTÉ AVB CI: A digital server-class
device with fixed I/O configuration of 12 mic/line
level inputs (with Sona AEC technology) and 8
mic/line level outputs, 4 GPIO connections, up
to 8 channels of configurable USB audio, and
networked audio via AVB.
TesiraFORTÉ AVB TI: A digital server-class device
with fixed I/O configuration of 12 mic/line level
inputs (with Sona AEC technology) and 8 mic/
line level outputs, 4 GPIO connections, up to 8
channels of configurable USB audio, 1 channel of
standard analog telephony, and networked audio
via AVB.
TesiraFORTÉ AVB VI: A digital server-class device
with fixed I/O configuration of 12 mic/line level
inputs (with Sona AEC technology) and 8 mic/
line level outputs, 4 GPIO connections, up to 8
channels of configurable USB audio, 2 channels
of VoIP telephony, and networked audio via AVB.
NON-AVB MODELS
TesiraFORTÉ AI: A digital server-class device with
fixed I/O configuration of 12 mic/line level inputs
and 8 mic/line level outputs, 4 GPIO connections,
and up to 8 channels of configurable USB audio.
TesiraFORTÉ CI: A digital server-class device with
fixed I/O configuration of 12 mic/line level inputs
(with Sona AEC technology) and 8 mic/line level
outputs, 4 GPIO connections, and up to
8 channels of configurable USB audio.
TesiraFORTÉ TI: A digital server-class device with
fixed I/O configuration of 12 mic/line level inputs
(with Sona AEC technology) and 8 mic/line level
outputs, 4 GPIO connections, up to 8 channels
of configurable USB audio, and 2 channels of
standard analog telephony.
TesiraFORTÉ VI: A digital server-class device with
fixed I/O configuration of 12 mic/line level inputs
(with Sona AEC technology) and 8 mic/line level
outputs, 4 GPIO connections, up to 8 channels of
configurable USB audio, and 2 channels of
VoIP telephony.
ETHERNET CONTROLS
TEC-1s: PoE Ethernet Control Surface Mount
TEC-1i: PoE Ethernet Control In-wall Mount
EXPANDERS
EX-IN: PoE+ 4-channel mic/line input expander
EX-AEC: PoE+ 4-channel mic/line input
expander with AEC
EX-OUT: PoE+ 4-channel mic/line output expander
EX-IO: PoE+ 2-channel mic/line input and
2-channel mic/line output expander
EX-MOD: Modular expander that can be configured
with up to three input and/or output cards
EX-LOGIC: PoE logic expander with 16 logic GPIO
(4 GPIO are configurable for potentiometer interface)
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15. Our Technical Support team has won accolades for the high level of service they provide
to our customers day in and day out. With unparalleled worldwide support, Biamp’s
applications engineers walk you through solutions, and assist you with new system designs,
programming, or troubleshooting.
They also perform our in-person and online equipment and concept trainings. From
webinars and tutorials on YouTube®
, to articles and tech notes on Cornerstone—our online
technical support knowledgebase, to in-person multi-day certification trainings, Biamp has
invested in your success.
WHEN YOU NEED
TRAINING & SUPPORT,
WE’VE GOT YOUR BACK.
TesiraFORTÉ is backed by
our 5-year warranty. Training is
available online and as part of
the in-person Tesira certification
training course. To learn more
about upcoming courses,
including schedules, go to
biamp.com/training.
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16. A: 9300 S.W. Gemini Drive
Beaverton, OR 97008 USA
T: 1.800.826.1457
T: +1 503.641.7287
E: biampinfo@biamp.com
W: http://biamp.com
CONTACT US