Telex Radio Dispatch manufactures and delivers thousands of IP-based mission critical communication systems worldwide, including their flagship C-Soft dispatch software which provides flexible and scalable dispatch capabilities. Telex converts audio and control functions to Ethernet packets allowing signals to be transported over IP networks for precise scaling of dispatch systems. The document provides an overview of Telex's IP-based radio dispatch solutions and components including consoles, interfaces, accessories, and software features.
Avaya VoIP on Cisco Best Practices by PacketBasePacketBase, Inc.
The document provides an overview of Avaya IP communications and best practices for interoperability with Cisco networks. It discusses key considerations for quality of service including recommended delay, jitter and packet loss thresholds. It also provides guidance on general QoS approaches, IP phone deployment, VLAN configuration, QoS settings for Cisco switches, and best practices for WAN connectivity.
BriCom provides software and hardware solutions that allow different radio systems and devices to interconnect over IP networks. This includes linking radio sites anywhere in the world and enabling legacy radios to connect to the internet. Their product line includes hardware devices that connect radios to IP networks and software applications that turn smartphones and computers into virtual radios. The solutions aim to improve communications for mobile workers and first responders.
The Clarity 4050 is an IP phone that functions as a 16-extension IP PBX embedded in a compact handset. It offers unified communications over fixed, mobile, and IP services using PacketRoute intelligent routing technology. PacketRoute allows multiple Clarity handsets to be connected to form a scalable hybrid IP PBX and intelligently routes calls between available SIM cards, PSTN trunks, and IP trunks. The Clarity 4050 provides flexibility through its routing plans and ability to reduce mobile call costs.
Reprinted with permission of NCTA, from the 2014 Cable Connection Spring Technical Forum Conference Proceedings. For more information on Cisco solutions, visit: http://www.cisco.com/c/en/us/solutions/index.html
Radio over IP (RoIP) is a technology that has emerged over the last 10 years and has provided a powerful and flexible mechanism for linking radios and dispatch consoles. RoIP has some inherent limitations and numerous technical challenges. This workshop will provide an overview of the technology, the key standards and some of the technical issues.
The workshop will cover on the following issues:
- The difference between RoIP and VoIP — how radio systems differ from phone systems
- Implementing one-to-one connections
- Implementing many-to-many connections
- Risk management: identifying network issues affecting RoIP/VoIP quality, maintenance and redundancy
- Design elements: building blocks; calculating network bandwidth requirements
The implications of RoIP for dispatch consoles will be also be discussed: how dispatch console to radio connections can be implemented with RoIP and how RoIP can be used to provide fault-tolerant dispatch architectures.
Finally, the workshop will look at the impact of new technologies such as IPv6, wireless broadband and the switch to digital radio on the RoIP landscape.
Paul Whitfield, Team Leader: Software Engineering, Omnitronics
The document proposes implementing Active Optical Network (AON) Fiber-To-The-X (FTTx) networks for VNPT to offer triple-play services, increase revenue, and compete effectively. It recommends broadband service tiers for business and home subscribers with throughput from 2Mbps to over 20Mbps, powered by VOLKTEK's market-proven FTTx switches, routers, and gateways. Ring topologies should serve high-ARPU business subscribers, while stars suffice for lower-ARPU homes to reduce costs.
We take pride in introducing ourselves to your esteemed organization. We are engaged in the business of providing Walkie Talkie and other Telecom and Security Solutions to our clients.
We are capable of giving all types of telecom spares with respect to Walkie Talkie and other Telecom and Security Products, viz: Telephone Handsets, Cords, Batteries, PTTs, Hand Held Metal Detector Chargers and Batteries, Door Frame Metal Detector Spares, etc. We do undertake maintenance contracts of various equipment with our staff of Technical Engineers and Technicians.
Avaya VoIP on Cisco Best Practices by PacketBasePacketBase, Inc.
The document provides an overview of Avaya IP communications and best practices for interoperability with Cisco networks. It discusses key considerations for quality of service including recommended delay, jitter and packet loss thresholds. It also provides guidance on general QoS approaches, IP phone deployment, VLAN configuration, QoS settings for Cisco switches, and best practices for WAN connectivity.
BriCom provides software and hardware solutions that allow different radio systems and devices to interconnect over IP networks. This includes linking radio sites anywhere in the world and enabling legacy radios to connect to the internet. Their product line includes hardware devices that connect radios to IP networks and software applications that turn smartphones and computers into virtual radios. The solutions aim to improve communications for mobile workers and first responders.
The Clarity 4050 is an IP phone that functions as a 16-extension IP PBX embedded in a compact handset. It offers unified communications over fixed, mobile, and IP services using PacketRoute intelligent routing technology. PacketRoute allows multiple Clarity handsets to be connected to form a scalable hybrid IP PBX and intelligently routes calls between available SIM cards, PSTN trunks, and IP trunks. The Clarity 4050 provides flexibility through its routing plans and ability to reduce mobile call costs.
Reprinted with permission of NCTA, from the 2014 Cable Connection Spring Technical Forum Conference Proceedings. For more information on Cisco solutions, visit: http://www.cisco.com/c/en/us/solutions/index.html
Radio over IP (RoIP) is a technology that has emerged over the last 10 years and has provided a powerful and flexible mechanism for linking radios and dispatch consoles. RoIP has some inherent limitations and numerous technical challenges. This workshop will provide an overview of the technology, the key standards and some of the technical issues.
The workshop will cover on the following issues:
- The difference between RoIP and VoIP — how radio systems differ from phone systems
- Implementing one-to-one connections
- Implementing many-to-many connections
- Risk management: identifying network issues affecting RoIP/VoIP quality, maintenance and redundancy
- Design elements: building blocks; calculating network bandwidth requirements
The implications of RoIP for dispatch consoles will be also be discussed: how dispatch console to radio connections can be implemented with RoIP and how RoIP can be used to provide fault-tolerant dispatch architectures.
Finally, the workshop will look at the impact of new technologies such as IPv6, wireless broadband and the switch to digital radio on the RoIP landscape.
Paul Whitfield, Team Leader: Software Engineering, Omnitronics
The document proposes implementing Active Optical Network (AON) Fiber-To-The-X (FTTx) networks for VNPT to offer triple-play services, increase revenue, and compete effectively. It recommends broadband service tiers for business and home subscribers with throughput from 2Mbps to over 20Mbps, powered by VOLKTEK's market-proven FTTx switches, routers, and gateways. Ring topologies should serve high-ARPU business subscribers, while stars suffice for lower-ARPU homes to reduce costs.
We take pride in introducing ourselves to your esteemed organization. We are engaged in the business of providing Walkie Talkie and other Telecom and Security Solutions to our clients.
We are capable of giving all types of telecom spares with respect to Walkie Talkie and other Telecom and Security Products, viz: Telephone Handsets, Cords, Batteries, PTTs, Hand Held Metal Detector Chargers and Batteries, Door Frame Metal Detector Spares, etc. We do undertake maintenance contracts of various equipment with our staff of Technical Engineers and Technicians.
This document discusses IPTV (Internet Protocol Television) and its key building blocks and protocols. IPTV allows distribution of TV content over IP networks and enables interactive viewing experiences. It uses H.264 video coding for high compression, supports multicast transmission, and requires a set-top box and broadband connection at the customer end. Key components include streaming servers, video servers, routers/switches, DSLAMs, and customer premises equipment. Quality of service is important for supporting real-time voice and video traffic.
Deploying Hybrid Local Area and Wide Area Video NetworksRonald Bartels
- The document discusses deploying hybrid local area network (LAN) and wide area network (WAN) video conferencing solutions. It describes using a Madge LAN Video Gateway to connect H.323 video codecs on the LAN to H.320 systems on the WAN and enable video calls between them. It also discusses using a WAN AccessSwitch to optimize WAN access costs and provide multipoint conferencing capabilities. Example network configurations are provided using these products to implement 128Kbps video conferencing between LAN and WAN locations.
This session will provide a quick review of the methodology of early dispatch systems connected to radio, telephone and other resources via circuit switched interfaces such as 4WE&M, 2W analogue etc., and their restricted backhaul capabilities, leading on to the 'stand-alone' RoIP boxes that allowed 4W E&M to be converted to IP and recovered at the other end allowing backhaul via more flexible IP networks.
The next technology is dispatch systems with native IP connectivity allowing the most flexible and functional interfaces between the dispatch system and its connected resources. While some manufacturers equipment uses proprietary IP messaging, most prefer and use open standards such as P25 CSSI (console sub system interface), DFSI (digital fixed system interface) and ISSI (inter sub system interface) or the emerging DMR AIS which ensures that different vendors equipment can interoperate with each other via these interfaces. Open standards provide end users with greatly improved competitive choice and functional capability on these systems.
The session will explore examples of IP interfaces for voice dispatch systems and the functions supported, plus give a background on how these apply to many different technologies and can even be adapted for conventional radio applications:
The workshop will cover on the following issues:
- The difference between RoIP and VoIP - how radio systems differ from phone systems
- Implementing one-to-one connections
- Implementing many-to-many connection
- Risk management: Identifying network issues affecting RoIP/VoIP quality; maintenance; and redundancy
- Design elements :- building blocks; calculating network bandwidth requirements
The implications of RoIP for dispatch consoles will be also be discussed: how dispatch console to radio connections can be implemented with RoIP and how RoIP can be used to provide fault tolerant dispatch architectures.
Finally the workshop will look at the impact of new technologies such as IPv6, Wireless Broadband and the switch to Digital Radio on the RoIP landscape.
Les Scott, Manager, System Sales, Zetron
The document discusses several ITU-T recommendations for telecommunication standards. It summarizes key recommendations for:
- IP frameworks (Y.1001)
- Digital subscriber lines (DSL), including ADSL, HDSL, SHDSL, and VDSL (G.990 series)
- Video and audio coding standards for multimedia communications, including H.261, H.263, H.264, and G.722 series.
The document discusses Alcatel-Lucent's Service Router Operating System (SR-OS) and their High Leverage Network (HLN). It introduces the HLN as a converged, scalable, and intelligent IP network that offers distributed service intelligence, broadband access, scalable IP transport, and supports innovative revenue-generating services. The HLN focuses on application enablement, universal access, network evolution, and operational transformation. It also discusses the FP3 chip that enables 100Gbps networking and the services supported on the HLN network.
A NEW SYSTEM ON CHIP RECONFIGURABLE GATEWAY ARCHITECTURE FOR VOICE OVER INTER...csandit
The aim of this paper is to present a new System on Chip (SoC) reconfigurable gateway
architecture for Voice over Internet Telephony (VOIP). Our motivation behind this work is
justified by the following arguments: most of VOIP solutions proposed in the market are based
on the use of a general purpose processor and a DSP circuit. In these solutions, the use of the
serial multiply accumulate circuit is very limiting for the signal processing. Also, in embedded
VOIP based DSP applications, the DSP works without MMU (memory management unit). This
is a serious limitation because VOIP solutions are multi-task based. In order to overcome these
problems, we propose a new VOIP gateway architecture built around the OpenRisc-1200-V3
processor. This last one integrates a DSP circuit as well as a MMU. The hardware architecture
is mapped into the VIRTEX-5 FPGA device. We propose a design methodology based on the
design for reuse and design with reuse concepts. We demonstrate that the proposed SoC
architecture is reconfigurable, scalable and the final RTL code can be reused for any FPGA or
ASIC technology. Performances measures, in the VIRTEX-5 FPGA device family, show that the
SOC-gateway architecture occupies 52% of the FPGA in term of slice LUT, 42% of IOBs, 60%
of bloc memory, 8% of integrated DSP, 16% of PLL and the total power is estimated at
4.3Watts.
The document discusses the drivers behind converging voice and data networks. The public switched telephone network (PSTN) architecture is inflexible and cannot adapt quickly enough, while data traffic has overtaken voice on many networks. True convergence of data, voice and video requires high-speed broadband access. IP networks provide a flexible, standards-based architecture and allow for more rapid feature development and deployment compared to the proprietary nature of the PSTN. Open interfaces in the packet, call control and application layers enable more vendors to create interoperable solutions.
Overview of VoIP (Voice over IP) and FoIP (Fax over IP) technologies like Session Initiation Protocol and H.323.
Even though voice over IP (VoIP) was hailed as a technological innovation, the idea to transport real-time traffic over TCP/IP networks was not new back in the 1990s when VoIP started being deployed in networks. Chapter 2.5 of the venerable RFC793 (TCP) shows both data oriented application traffic as well as voice being transported over IP based networks.
Nevertheless, VoIP puts high demands on signal and protocol processing capabilities so it became possible at reasonable costs only in the 1990s.
VoIP can be roughly split into two main functions. Signaling protocols like SIP (Session Initiation Protocol), H.323 and MGCP/H.248 are used to establish a conference session and the data path for transporting real-time voice data packets. SIP has largely supplanted H.323 in recent years to its simpler structure and packet sequences. MGCP and H.248 are mostly used in carrier backbone networks.
Protocols like RTP (Real Time Protocol) transport voice packets and provide the necessary information for receivers to equalize packet flow variations to provide a smooth playback of the original voice signal.
Voice codecs are one of the core functions of the data path. Voice compression reduces the bandwidth required to transport voice over an IP based network. Compression may be less of a concern in local area networks with gigabit speeds, on slower links like 3G (UMTS, LTE) it still makes a lot of sense.
The algorithms used in different codecs make use of various characteristics of the characteristics of human speech recognition. Redundant information is removed from the signals thus slightly reducing the quality, but greatly reducing the required bandwidth.
In VoIP networks, the echo problem is typically compounded by the increased delay incurred by packetization of voice signals. To counteract the echo problem, VoIP gear (hard phones, soft phones, gateways) include echo cancelers to remove echo signals from the transmit signal.
To transport facsimile over an IP based network, even more technology is needed. Facsimile protocols are very susceptible to delay and delay variation and thus need more compensation algorithms. Protocols like T.38 terminate facsimile protocols like T.30 (analog facsimile) and transport the fax images as digitized pictures over IP based networks.
Efficient Telecommunication Infrastructure with Internet Telephony (VoIP)Thomas Siegers
This document provides an overview of efficient telecommunication infrastructure using Internet telephony (VoIP). It discusses basics of telephony and networking, protocols like SIP and RTP, VoIP hardware, service providers, and examples of integrating VoIP into networks and PBX systems. Skype and issues with it are also covered. The presentation aims to explain how VoIP works and scenarios for implementing efficient VoIP solutions.
Surf Solutions intro to new partner jan2011 Rudy Shainer
This document discusses Surf's enabling technology products for multimedia applications and services. It introduces Surf's two product lines - enabling technology and complete "box" solutions. The enabling technology line includes boards and mezzanine cards with video/audio/voice codecs and modems. These include the SURFAce-812 PCIe card with 8 DSPs, the SURFExpress PCIe board, and the SURFRider AMC mezzanine card. These products offer modular designs using Surf's SURFDocker architecture and are aimed at OEM customers and systems integrators to enable high capacity multimedia applications.
El camino hacia la 5G - Isidro Nieto from Ericsson at VI Jornadas Cátedras Te...Ericsson España
Participation of Isidro Nieto, Regional Head of Mobile Broadband for Ericsson in the Mediterranean, during the event VI Jornadas Cátedras Telefónica UPM, Madrid.
VoIP allows users to make phone calls using an internet connection instead of a regular phone line. It works by converting voice signals from phones into digital data packets that can be transmitted over the internet or broadband connection. Common VoIP protocols include SIP, H.323, and IAX. VoIP has benefits like lower call costs, integration of voice and data, and added features like video calling. It has applications in agriculture by providing communication for remote farmers through initiatives like the Caribbean Farmers Network and the Timbaktu Collective project in India.
The document discusses technologies relevant to voice over IP (VoIP) applications including voice processing modules, codecs, signaling protocols, transport protocols, and network traversal techniques. It also covers business models, community aspects, and opportunities in premium services and bridging across technologies and communities. Emerging areas discussed include VoIP on mobile networks and the roles of portals, social networks, and device vendors.
Telcos are using IP networks and advanced video encoding like MPEG-4 Part 10 (H.264) to deliver video content over the internet via IPTV. H.264 provides much more bandwidth efficiency than MPEG-2. VoIP maps the encoded video onto the IP network for transmission. A typical implementation includes an H.264 encoder, VoIP to encapsulate the stream, and transmission over Ethernet. IPTV allows telcos to compete with cable companies by providing digital TV using internet infrastructure instead of traditional cable networks.
This document provides an overview of Voice over IP (VoIP) technology. It explains that VoIP allows voice calls to be made over an IP network like the internet rather than the traditional public switched telephone network. It describes how VoIP works by converting voice signals to digital form and transmitting them in packets over the network. It also outlines the key components of a VoIP system including protocols, gateways, codecs and the network architecture. Additionally, it discusses benefits of VoIP like reduced communication costs and better utilization of bandwidth.
This document discusses the emergence of system-on-chip (SOC) semiconductors to address the need for highly integrated, high-density, and low-power voice-over-packet (VoP) equipment solutions. It describes how SOC-based VoP processors integrate multiple digital signal processors, microprocessors, memory and other resources onto a single chip. This allows for solutions that are more scalable, flexible and energy efficient than traditional discrete component-based VoP modules.
The Voice Evolution—VoLTE. VoHSPA+, WCDMA+ and QualityQualcomm Research
The wireless industry is moving toward a richer, packet based voice on IP. However, traditional mobile Circuit Switched (CS) voice has a long life during the transition to carrier grade VoIP. Qualcomm’s comprehensive voice evolution technology leadership provides all aspects needed for the successful transition to carrier grade VoIP: We make the 3G CS voice more efficient (1X Advanced and WCDMA+), and lead the mobile VoIP delivery technologies with IMS VoIP over LTE (VoLTE) and HSPA+, provide the necessary transitions technologies between circuit switched voice and IMS VoIP; And finally, we provide the voice quality technologies to ensure the best voice experience for 3G CS voice, IMS VoIP as well as OTT VoIP.
For more information please visit www.qualcomm.com/wcdma
Download the presentation here: http://www.qualcomm.com/media/documents/voice-evolution-volte-vohspa-wcdma-and-quality
Mobile Transport Evolution with Unified MPLSCisco Canada
Mobile Service Providers are seeing unprecedented challenges in relation to their Transport architectures with the 3GPP evolution towards IP based Node Bs, LTE (Long Term Evolution) and LTE-Advanced. This presentation will initially discuss the network migration trends and factors that are changing how mobile networks are evolving. A description is provided on Unified MPLS and the current issues that need to be fixed and how this architecture addresses this. A more detailed analysis will then examine the options available for transporting GSM/2G, UMTS/3G traffic and IP/Ethernet Node B deployments and some of factors that need consideration like scalability, resiliency and security. Finally, there is a detailed description of the LTE/LTE - A evolution and the feature requirements made on the transport network. There will be detailed analysis of different LTE models and also some technical enhancements and proposals considered for the implementation of LTE in a Unified MPLS environment.
This document discusses and compares VoIP (Voice over Internet Protocol) and PSTN (Public Switched Telephone Network) networks. It provides details on the evolution and architecture of PSTN networks and how they facilitate voice communications. It also describes how IP networks were developed for higher speed data and internet access. While most telecom providers built separate networks for voice and data, some now offer VoIP over their IP networks. The document analyzes advantages and disadvantages of carrying voice calls over IP networks versus circuit-switched networks like PSTN. It argues that a mixed network approach using both is most cost effective and ensures quality for real-time communications like voice and video.
The document discusses the evolution of next generation IP transport networks using a Unified MPLS approach. Key points include:
- Unified MPLS aims to simplify MPLS operations in large, complex networks through common MPLS technology across domains and hierarchical BGP routing.
- It reduces the number of operational points needed for services by minimizing static configuration and integrating previously separate MPLS islands.
- The network is divided into IGP/LDP domains with inter-domain communication handled through labeled BGP routes. This reduces route tables and the number of label switched paths required in the access domains.
Discussion on IoT technologies – CAT M1 and NB-IoT (CAT M2)Small Cell Forum
This document discusses CAT M1 and NB-IoT low power wide area network technologies for IoT. It provides an overview of the technical capabilities and differences between CAT M1 and NB-IoT, such as peak data rates, antenna configurations, mobility support and spectrum usage. It also discusses early deployments by mobile operators and trials underway. Radisys positioning of CAT M1 and NB-IoT products is presented, with plans for interoperability demonstrations.
Lamit Company offers satellite communication services and equipment including broadband satellite internet, virtual private networks, video surveillance, and voice over IP solutions. They have designed their own software and hardware platforms to support present and future networks for applications such as broadcast, point-to-point, and point-to-multipoint transmissions for commercial, government, and military use. Lamit also provides consultancy services for business, management, and IT and telecommunications technical issues.
This document discusses IPTV (Internet Protocol Television) and its key building blocks and protocols. IPTV allows distribution of TV content over IP networks and enables interactive viewing experiences. It uses H.264 video coding for high compression, supports multicast transmission, and requires a set-top box and broadband connection at the customer end. Key components include streaming servers, video servers, routers/switches, DSLAMs, and customer premises equipment. Quality of service is important for supporting real-time voice and video traffic.
Deploying Hybrid Local Area and Wide Area Video NetworksRonald Bartels
- The document discusses deploying hybrid local area network (LAN) and wide area network (WAN) video conferencing solutions. It describes using a Madge LAN Video Gateway to connect H.323 video codecs on the LAN to H.320 systems on the WAN and enable video calls between them. It also discusses using a WAN AccessSwitch to optimize WAN access costs and provide multipoint conferencing capabilities. Example network configurations are provided using these products to implement 128Kbps video conferencing between LAN and WAN locations.
This session will provide a quick review of the methodology of early dispatch systems connected to radio, telephone and other resources via circuit switched interfaces such as 4WE&M, 2W analogue etc., and their restricted backhaul capabilities, leading on to the 'stand-alone' RoIP boxes that allowed 4W E&M to be converted to IP and recovered at the other end allowing backhaul via more flexible IP networks.
The next technology is dispatch systems with native IP connectivity allowing the most flexible and functional interfaces between the dispatch system and its connected resources. While some manufacturers equipment uses proprietary IP messaging, most prefer and use open standards such as P25 CSSI (console sub system interface), DFSI (digital fixed system interface) and ISSI (inter sub system interface) or the emerging DMR AIS which ensures that different vendors equipment can interoperate with each other via these interfaces. Open standards provide end users with greatly improved competitive choice and functional capability on these systems.
The session will explore examples of IP interfaces for voice dispatch systems and the functions supported, plus give a background on how these apply to many different technologies and can even be adapted for conventional radio applications:
The workshop will cover on the following issues:
- The difference between RoIP and VoIP - how radio systems differ from phone systems
- Implementing one-to-one connections
- Implementing many-to-many connection
- Risk management: Identifying network issues affecting RoIP/VoIP quality; maintenance; and redundancy
- Design elements :- building blocks; calculating network bandwidth requirements
The implications of RoIP for dispatch consoles will be also be discussed: how dispatch console to radio connections can be implemented with RoIP and how RoIP can be used to provide fault tolerant dispatch architectures.
Finally the workshop will look at the impact of new technologies such as IPv6, Wireless Broadband and the switch to Digital Radio on the RoIP landscape.
Les Scott, Manager, System Sales, Zetron
The document discusses several ITU-T recommendations for telecommunication standards. It summarizes key recommendations for:
- IP frameworks (Y.1001)
- Digital subscriber lines (DSL), including ADSL, HDSL, SHDSL, and VDSL (G.990 series)
- Video and audio coding standards for multimedia communications, including H.261, H.263, H.264, and G.722 series.
The document discusses Alcatel-Lucent's Service Router Operating System (SR-OS) and their High Leverage Network (HLN). It introduces the HLN as a converged, scalable, and intelligent IP network that offers distributed service intelligence, broadband access, scalable IP transport, and supports innovative revenue-generating services. The HLN focuses on application enablement, universal access, network evolution, and operational transformation. It also discusses the FP3 chip that enables 100Gbps networking and the services supported on the HLN network.
A NEW SYSTEM ON CHIP RECONFIGURABLE GATEWAY ARCHITECTURE FOR VOICE OVER INTER...csandit
The aim of this paper is to present a new System on Chip (SoC) reconfigurable gateway
architecture for Voice over Internet Telephony (VOIP). Our motivation behind this work is
justified by the following arguments: most of VOIP solutions proposed in the market are based
on the use of a general purpose processor and a DSP circuit. In these solutions, the use of the
serial multiply accumulate circuit is very limiting for the signal processing. Also, in embedded
VOIP based DSP applications, the DSP works without MMU (memory management unit). This
is a serious limitation because VOIP solutions are multi-task based. In order to overcome these
problems, we propose a new VOIP gateway architecture built around the OpenRisc-1200-V3
processor. This last one integrates a DSP circuit as well as a MMU. The hardware architecture
is mapped into the VIRTEX-5 FPGA device. We propose a design methodology based on the
design for reuse and design with reuse concepts. We demonstrate that the proposed SoC
architecture is reconfigurable, scalable and the final RTL code can be reused for any FPGA or
ASIC technology. Performances measures, in the VIRTEX-5 FPGA device family, show that the
SOC-gateway architecture occupies 52% of the FPGA in term of slice LUT, 42% of IOBs, 60%
of bloc memory, 8% of integrated DSP, 16% of PLL and the total power is estimated at
4.3Watts.
The document discusses the drivers behind converging voice and data networks. The public switched telephone network (PSTN) architecture is inflexible and cannot adapt quickly enough, while data traffic has overtaken voice on many networks. True convergence of data, voice and video requires high-speed broadband access. IP networks provide a flexible, standards-based architecture and allow for more rapid feature development and deployment compared to the proprietary nature of the PSTN. Open interfaces in the packet, call control and application layers enable more vendors to create interoperable solutions.
Overview of VoIP (Voice over IP) and FoIP (Fax over IP) technologies like Session Initiation Protocol and H.323.
Even though voice over IP (VoIP) was hailed as a technological innovation, the idea to transport real-time traffic over TCP/IP networks was not new back in the 1990s when VoIP started being deployed in networks. Chapter 2.5 of the venerable RFC793 (TCP) shows both data oriented application traffic as well as voice being transported over IP based networks.
Nevertheless, VoIP puts high demands on signal and protocol processing capabilities so it became possible at reasonable costs only in the 1990s.
VoIP can be roughly split into two main functions. Signaling protocols like SIP (Session Initiation Protocol), H.323 and MGCP/H.248 are used to establish a conference session and the data path for transporting real-time voice data packets. SIP has largely supplanted H.323 in recent years to its simpler structure and packet sequences. MGCP and H.248 are mostly used in carrier backbone networks.
Protocols like RTP (Real Time Protocol) transport voice packets and provide the necessary information for receivers to equalize packet flow variations to provide a smooth playback of the original voice signal.
Voice codecs are one of the core functions of the data path. Voice compression reduces the bandwidth required to transport voice over an IP based network. Compression may be less of a concern in local area networks with gigabit speeds, on slower links like 3G (UMTS, LTE) it still makes a lot of sense.
The algorithms used in different codecs make use of various characteristics of the characteristics of human speech recognition. Redundant information is removed from the signals thus slightly reducing the quality, but greatly reducing the required bandwidth.
In VoIP networks, the echo problem is typically compounded by the increased delay incurred by packetization of voice signals. To counteract the echo problem, VoIP gear (hard phones, soft phones, gateways) include echo cancelers to remove echo signals from the transmit signal.
To transport facsimile over an IP based network, even more technology is needed. Facsimile protocols are very susceptible to delay and delay variation and thus need more compensation algorithms. Protocols like T.38 terminate facsimile protocols like T.30 (analog facsimile) and transport the fax images as digitized pictures over IP based networks.
Efficient Telecommunication Infrastructure with Internet Telephony (VoIP)Thomas Siegers
This document provides an overview of efficient telecommunication infrastructure using Internet telephony (VoIP). It discusses basics of telephony and networking, protocols like SIP and RTP, VoIP hardware, service providers, and examples of integrating VoIP into networks and PBX systems. Skype and issues with it are also covered. The presentation aims to explain how VoIP works and scenarios for implementing efficient VoIP solutions.
Surf Solutions intro to new partner jan2011 Rudy Shainer
This document discusses Surf's enabling technology products for multimedia applications and services. It introduces Surf's two product lines - enabling technology and complete "box" solutions. The enabling technology line includes boards and mezzanine cards with video/audio/voice codecs and modems. These include the SURFAce-812 PCIe card with 8 DSPs, the SURFExpress PCIe board, and the SURFRider AMC mezzanine card. These products offer modular designs using Surf's SURFDocker architecture and are aimed at OEM customers and systems integrators to enable high capacity multimedia applications.
El camino hacia la 5G - Isidro Nieto from Ericsson at VI Jornadas Cátedras Te...Ericsson España
Participation of Isidro Nieto, Regional Head of Mobile Broadband for Ericsson in the Mediterranean, during the event VI Jornadas Cátedras Telefónica UPM, Madrid.
VoIP allows users to make phone calls using an internet connection instead of a regular phone line. It works by converting voice signals from phones into digital data packets that can be transmitted over the internet or broadband connection. Common VoIP protocols include SIP, H.323, and IAX. VoIP has benefits like lower call costs, integration of voice and data, and added features like video calling. It has applications in agriculture by providing communication for remote farmers through initiatives like the Caribbean Farmers Network and the Timbaktu Collective project in India.
The document discusses technologies relevant to voice over IP (VoIP) applications including voice processing modules, codecs, signaling protocols, transport protocols, and network traversal techniques. It also covers business models, community aspects, and opportunities in premium services and bridging across technologies and communities. Emerging areas discussed include VoIP on mobile networks and the roles of portals, social networks, and device vendors.
Telcos are using IP networks and advanced video encoding like MPEG-4 Part 10 (H.264) to deliver video content over the internet via IPTV. H.264 provides much more bandwidth efficiency than MPEG-2. VoIP maps the encoded video onto the IP network for transmission. A typical implementation includes an H.264 encoder, VoIP to encapsulate the stream, and transmission over Ethernet. IPTV allows telcos to compete with cable companies by providing digital TV using internet infrastructure instead of traditional cable networks.
This document provides an overview of Voice over IP (VoIP) technology. It explains that VoIP allows voice calls to be made over an IP network like the internet rather than the traditional public switched telephone network. It describes how VoIP works by converting voice signals to digital form and transmitting them in packets over the network. It also outlines the key components of a VoIP system including protocols, gateways, codecs and the network architecture. Additionally, it discusses benefits of VoIP like reduced communication costs and better utilization of bandwidth.
This document discusses the emergence of system-on-chip (SOC) semiconductors to address the need for highly integrated, high-density, and low-power voice-over-packet (VoP) equipment solutions. It describes how SOC-based VoP processors integrate multiple digital signal processors, microprocessors, memory and other resources onto a single chip. This allows for solutions that are more scalable, flexible and energy efficient than traditional discrete component-based VoP modules.
The Voice Evolution—VoLTE. VoHSPA+, WCDMA+ and QualityQualcomm Research
The wireless industry is moving toward a richer, packet based voice on IP. However, traditional mobile Circuit Switched (CS) voice has a long life during the transition to carrier grade VoIP. Qualcomm’s comprehensive voice evolution technology leadership provides all aspects needed for the successful transition to carrier grade VoIP: We make the 3G CS voice more efficient (1X Advanced and WCDMA+), and lead the mobile VoIP delivery technologies with IMS VoIP over LTE (VoLTE) and HSPA+, provide the necessary transitions technologies between circuit switched voice and IMS VoIP; And finally, we provide the voice quality technologies to ensure the best voice experience for 3G CS voice, IMS VoIP as well as OTT VoIP.
For more information please visit www.qualcomm.com/wcdma
Download the presentation here: http://www.qualcomm.com/media/documents/voice-evolution-volte-vohspa-wcdma-and-quality
Mobile Transport Evolution with Unified MPLSCisco Canada
Mobile Service Providers are seeing unprecedented challenges in relation to their Transport architectures with the 3GPP evolution towards IP based Node Bs, LTE (Long Term Evolution) and LTE-Advanced. This presentation will initially discuss the network migration trends and factors that are changing how mobile networks are evolving. A description is provided on Unified MPLS and the current issues that need to be fixed and how this architecture addresses this. A more detailed analysis will then examine the options available for transporting GSM/2G, UMTS/3G traffic and IP/Ethernet Node B deployments and some of factors that need consideration like scalability, resiliency and security. Finally, there is a detailed description of the LTE/LTE - A evolution and the feature requirements made on the transport network. There will be detailed analysis of different LTE models and also some technical enhancements and proposals considered for the implementation of LTE in a Unified MPLS environment.
This document discusses and compares VoIP (Voice over Internet Protocol) and PSTN (Public Switched Telephone Network) networks. It provides details on the evolution and architecture of PSTN networks and how they facilitate voice communications. It also describes how IP networks were developed for higher speed data and internet access. While most telecom providers built separate networks for voice and data, some now offer VoIP over their IP networks. The document analyzes advantages and disadvantages of carrying voice calls over IP networks versus circuit-switched networks like PSTN. It argues that a mixed network approach using both is most cost effective and ensures quality for real-time communications like voice and video.
The document discusses the evolution of next generation IP transport networks using a Unified MPLS approach. Key points include:
- Unified MPLS aims to simplify MPLS operations in large, complex networks through common MPLS technology across domains and hierarchical BGP routing.
- It reduces the number of operational points needed for services by minimizing static configuration and integrating previously separate MPLS islands.
- The network is divided into IGP/LDP domains with inter-domain communication handled through labeled BGP routes. This reduces route tables and the number of label switched paths required in the access domains.
Discussion on IoT technologies – CAT M1 and NB-IoT (CAT M2)Small Cell Forum
This document discusses CAT M1 and NB-IoT low power wide area network technologies for IoT. It provides an overview of the technical capabilities and differences between CAT M1 and NB-IoT, such as peak data rates, antenna configurations, mobility support and spectrum usage. It also discusses early deployments by mobile operators and trials underway. Radisys positioning of CAT M1 and NB-IoT products is presented, with plans for interoperability demonstrations.
Lamit Company offers satellite communication services and equipment including broadband satellite internet, virtual private networks, video surveillance, and voice over IP solutions. They have designed their own software and hardware platforms to support present and future networks for applications such as broadcast, point-to-point, and point-to-multipoint transmissions for commercial, government, and military use. Lamit also provides consultancy services for business, management, and IT and telecommunications technical issues.
Albedo Telecom offers a range of telecom products and services to help clients evolve their telecom investments, including network design, IPTV, VoIP, testing equipment, emulation of network conditions, and professional training. They provide solutions for migration to carrier Ethernet, service level agreement monitoring and verification, and interoperability testing to ensure equipment meets business requirements. Albedo is a global supplier with over 25 years of experience in telecom technologies and develops complete telecom projects from design to manufacturing.
This document discusses the drivers behind converging voice and data networking. It describes drawbacks of the public switched telephone network (PSTN) including its inability to quickly deploy new features or converge data, voice, and video. The document outlines how IP networking provides a more flexible architecture and open standards to enable this convergence compared to the proprietary nature of the PSTN. Key components that enable voice over IP like RTP, call control protocols, and an open application layer are also summarized.
This document provides an overview of Sigfox and its capabilities. Sigfox is a global LPWA network that provides low power, global connectivity at a low cost to address a variety of use cases. It discusses Sigfox's network coverage, market size, and applications such as asset tracking, smart city solutions, and home security. The document also covers Sigfox's certification process, development tools, and resources to help developers build connected devices and solutions on the Sigfox network.
Digital Satellites is a telecom systems integrator established in 2011 in Egypt that provides telecommunication services and solutions primarily to the oil, gas, and energy sectors. It has 44 permanent staff and a cumulative project volume of $23 million. The company offers advanced technology and telecommunication solutions including networks, voice and data services, and safety and security solutions. Digital Satellites also provides satellite communication services through teleports located in the UK, Egypt, and Saudi Arabia.
The document discusses several topics related to digital networks and protocols:
1. TCP/IP (Transmission Control Protocol/Internet Protocol) is an important protocol suite that defines how devices communicate over the internet through a layered model.
2. Circuit-switched services determine the optimal network path for transferring application data based on the requested quality of service.
3. Digital media adapters like the ZyXEL DMA-1000 allow users to stream computer files to televisions and stereo systems over a local area network or USB connection.
The document discusses several topics related to digital networks and protocols:
1. TCP/IP (Transmission Control Protocol/Internet Protocol) is an important protocol suite that defines how devices communicate over the internet through a layered model.
2. Circuit-switched services determine the optimal network path for transferring application data based on the requested quality of service.
3. Digital media adapters like the ZyXEL DMA-1000 allow users to stream computer files to televisions and stereo systems over a local area network or USB connection.
A NEW SYSTEM ON CHIP RECONFIGURABLE GATEWAY ARCHITECTURE FOR VOICE OVER INTER...cscpconf
The aim of this paper is to present a new System on Chip (SoC) reconfigurable gateway architecture for Voice over Internet Telephony (VOIP). Our motivation behind this work is
justified by the following arguments: most of VOIP solutions proposed in the market are based on the use of a general purpose processor and a DSP circuit. In these solutions, the use of the serial multiply accumulate circuit is very limiting for the signal processing. Also, in embedded VOIP based DSP applications, the DSP works without MMU (memory management unit). This is a serious limitation because VOIP solutions are multi-task based. In order to overcome these
problems, we propose a new VOIP gateway architecture built around the OpenRisc-1200-V3 processor. This last one integrates a DSP circuit as well as a MMU. The hardware architecture is mapped into the VIRTEX-5 FPGA device. We propose a design methodology based on the design for reuse and design with reuse concepts. We demonstrate that the proposed SoC architecture is reconfigurable, scalable and the final RTL code can be reused for any FPGA or ASIC technology. Performances measures, in the VIRTEX-5 FPGA device family, show that the SOC-gateway architecture occupies 52% of the FPGA in term of slice LUT, 42% of IOBs, 60% of bloc memory, 8% of integrated DSP, 16% of PLL and the total power is estimated at 4.3Watts
The Core CPE team at Promwad consists of 8 software engineers, 3 hardware engineers, and 3 QA engineers with over 45 years of combined software experience. The team develops CPE devices using SoCs from manufacturers like Marvell and Realtek running software like OpenWRT. Key areas of focus include VoIP, DECT, wireless, and remote management via standards like TR-069.
An increasing amount of TV channels, transmitted at higher
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The document discusses Voice over IP (VoIP) and IP telephony. It explains that VoIP allows phone calls to be made over an IP network like the internet instead of the traditional public switched telephone network (PSTN). VoIP offers cost savings compared to PSTN and enables additional features like video calls and mobility. The document also discusses when companies should consider replacing their private branch exchange (PBX) phone system with an IP telephony system using VoIP.
Radisys Virtualized RAN using the Mobile-CORD platformSmall Cell Forum
The document discusses Radisys' virtualized RAN solution using the Mobile-CORD platform. It provides an overview of the lab setup used including Intel FlexRAN, Radisys EPC, and third party RRH. Features like single cell operation, carrier aggregation, and 4x4 MIMO are supported. The solution deploys the virtualized RAN functions on the M-CORD platform using Radisys DCEngine hardware. Next steps include integrating with ONAP/ECOMP and demonstrating use cases like mobile core network CO as datacenter and eNodeB local breakout.
The document discusses TCP/IP protocols, circuit-switched services, digital access equipment, and receiver apparatuses. TCP/IP is an important internet protocol suite consisting of four layers. Circuit-switched services determine the optimal network for transferring application flows. Digital access equipment like the ZyXEL DMA-1000 allows transferring computer files to televisions and stereos. Receiver apparatuses have many branches selected through earth connections of coaxial cables.
ViaSat is a satellite communications company founded in 1986 that provides broadband internet services using its own satellites and technology. It has developed innovative satellite networking solutions like LinkStarS2, Skylinx, and ArcLight that deliver broadband internet to customers in rural and remote areas at competitive prices. ViaSat also provides tools to optimize network performance and accelerate application delivery to improve the customer experience.
Vinay Tripathi has over 13 years of experience in software development for network and telecom products. He currently works as a Senior Technical Architect at HCL Technologies in Bangalore. Some of his areas of expertise include datacom devices, broadband routers, IP phones, video conferencing, DOCSIS standards, Wi-Fi, routing protocols, Linux, and network security. He has extensive experience leading projects for clients such as Ericsson, Cisco, Technicolor, and Broadcom.
IP telephony has received interest from many users and organizations as it provides cost savings over traditional phone lines. VoIP saves money by using existing computer networks and IP infrastructure rather than separate phone lines, reducing line charges, feature charges, taxes, and fees. Many organizations currently maintain separate networks for data and voice, but integrating the two using VoIP provides a more cost effective and flexible unified solution.
Ethernet was invented in the 1970s at Xerox PARC and was later commercialized. It is a widely used wired networking technology that uses bus topology and CSMA/CD protocol to allow multiple devices to share bandwidth on the same network. Li-Fi is a new wireless technology that uses visible light communication through LED lights to transmit data, providing a potential alternative to Wi-Fi that has benefits like higher speeds, more bandwidth availability, and better security. It was introduced in 2011 and companies are working to commercialize Li-Fi products and networks. Potential applications include use in places where radio signals cannot be used safely or are restricted.
Cambium Networks is an industry leader in point-to-multipoint and point-to-point wireless broadband solutions. They have shipped over 4 million nodes totaling over $1 billion to networks in more than 150 countries. Their ePMP product line provides affordable and scalable wireless access networks through features like GPS synchronization, high scalability and consistent performance, interference mitigation technology, and effective quality of service capabilities.
The VX 9000 virtualized software-based wireless LAN controller combines the power of virtualization with Motorola Solutions' WiNG Controller. It provides centralized management of wireless networks through a single interface with high scalability, flexibility and advanced wireless services. Key features include integrated network security, the advanced WiNG 5 operating system, plug-and-play deployment, simplified licensing and infinite scalability through virtualization. It supports all major hypervisors and public/private clouds for maximum deployment flexibility at low cost.
The NX 7500 integrated services platform provides comprehensive management of up to 2,048 network elements through a single interface. It allows all network infrastructure to intelligently route traffic for maximum speed and throughput without congestion. The NX 7500 offers advanced wireless LAN performance for mid-sized and campus environments with features such as plug-and-play installation, hierarchical management, smart routing, BYOD support, and integrated security services. It provides flexibility and investment protection through modular upgrades.
The document discusses the challenges retailers face in supporting increased wireless applications and next-generation Wi-Fi in stores. It introduces the Motorola AP 8200 Series as a solution that provides high-performance wireless connectivity for customers and staff. The AP 8200 Series allows easy access, security, support for 802.11ac Wi-Fi, bandwidth for applications, and performance for many users. It provides flexibility, a cost-effective upgrade to 802.11ac, and features for security, environmental monitoring, location services, and more.
The document describes the innovative features of the Motorola AP 8222 wireless access point. It has a sleek design suitable for retail, office, and other customer-facing spaces. It provides dual-band 802.11ac and 802.11n wireless connectivity at speeds up to 1.3Gbps. Key features include advanced beamforming, gap-free security, and support for bandwidth-heavy applications like video calling. The access point is centrally managed through Motorola's WiNG 5 networking operating system.
The AP 8163 is a ruggedized outdoor mesh access point designed to withstand extreme weather conditions. It has three radios - two for client access across 2.4GHz and 5GHz bands, and a third radio that can be used for wireless intrusion prevention scanning or dynamic frequency selection to avoid radar interference. The advanced WiNG 5 operating system allows the access points to self-optimize the network for best performance. Key features include powerful antennas for extended range, mesh networking for redundancy, and security features like firewalls and wireless intrusion prevention.
The document describes the features and capabilities of the Motorola AP 8122 3x3 MIMO 802.11n access point. It delivers high throughput to support enterprise applications including voice and HD video using 802.11n technology with standard 802.3af PoE. It has advanced features like load balancing, pre-emptive roaming, and dual band radios to increase network reliability, resilience, and security. The access point also supports advanced wireless capabilities such as voice over wireless, location services, and guest access controls.
The document describes the innovative features of the AP 7532 wireless access point. It provides the highest wireless speeds available with 3x3 MIMO and 256 QAM modulation on both 2.4GHz and 5GHz radios. It has a dual radio 802.11ac/802.11n design that provides a upgrade path to 1.3Gbps 802.11ac speeds while maintaining support for existing devices. It offers various advanced features like load balancing, security, sensor support and quality of service for voice. The access point is designed to deliver maximum performance at a low cost.
The AP 7502 is a dual-band 802.11ac wireless access point designed for installation in small spaces like hotel rooms. It has a compact wall-mount design, supports the latest WiFi standards, and includes features to ensure reliable connectivity even in challenging environments. Setup and management are simplified through zero-touch provisioning and both standalone and controller-based operation modes.
The document describes the innovative features of the AP 7522 wireless access point. It provides dual-band 802.11ac and 802.11n radios for high performance WiFi. It offers internal or external antenna options and can function as both an access point and wireless sensor. The access point provides security, load balancing, and other features to support mission critical applications on the wireless network.
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Understanding User Needs and Satisfying ThemAggregage
https://www.productmanagementtoday.com/frs/26903918/understanding-user-needs-and-satisfying-them
We know we want to create products which our customers find to be valuable. Whether we label it as customer-centric or product-led depends on how long we've been doing product management. There are three challenges we face when doing this. The obvious challenge is figuring out what our users need; the non-obvious challenges are in creating a shared understanding of those needs and in sensing if what we're doing is meeting those needs.
In this webinar, we won't focus on the research methods for discovering user-needs. We will focus on synthesis of the needs we discover, communication and alignment tools, and how we operationalize addressing those needs.
Industry expert Scott Sehlhorst will:
• Introduce a taxonomy for user goals with real world examples
• Present the Onion Diagram, a tool for contextualizing task-level goals
• Illustrate how customer journey maps capture activity-level and task-level goals
• Demonstrate the best approach to selection and prioritization of user-goals to address
• Highlight the crucial benchmarks, observable changes, in ensuring fulfillment of customer needs
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Building Your Employer Brand with Social MediaLuanWise
Presented at The Global HR Summit, 6th June 2024
In this keynote, Luan Wise will provide invaluable insights to elevate your employer brand on social media platforms including LinkedIn, Facebook, Instagram, X (formerly Twitter) and TikTok. You'll learn how compelling content can authentically showcase your company culture, values, and employee experiences to support your talent acquisition and retention objectives. Additionally, you'll understand the power of employee advocacy to amplify reach and engagement – helping to position your organization as an employer of choice in today's competitive talent landscape.
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How to Start Up a Company: A Step-by-Step Guide Starting a company is an exciting adventure that combines creativity, strategy, and hard work. It can seem overwhelming at first, but with the right guidance, anyone can transform a great idea into a successful business. Let's dive into how to start up a company, from the initial spark of an idea to securing funding and launching your startup.
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This PowerPoint compilation offers a comprehensive overview of 20 leading innovation management frameworks and methodologies, selected for their broad applicability across various industries and organizational contexts. These frameworks are valuable resources for a wide range of users, including business professionals, educators, and consultants.
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INCLUDED FRAMEWORKS/MODELS:
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2. IDEO’s Human-Centered Design
3. Strategyzer’s Business Model Innovation
4. Lean Startup Methodology
5. Agile Innovation Framework
6. Doblin’s Ten Types of Innovation
7. McKinsey’s Three Horizons of Growth
8. Customer Journey Map
9. Christensen’s Disruptive Innovation Theory
10. Blue Ocean Strategy
11. Strategyn’s Jobs-To-Be-Done (JTBD) Framework with Job Map
12. Design Sprint Framework
13. The Double Diamond
14. Lean Six Sigma DMAIC
15. TRIZ Problem-Solving Framework
16. Edward de Bono’s Six Thinking Hats
17. Stage-Gate Model
18. Toyota’s Six Steps of Kaizen
19. Microsoft’s Digital Transformation Framework
20. Design for Six Sigma (DFSS)
To download this presentation, visit:
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2. More installations than any other IP-based dispatch system.
Table of Contents
Features:
IP-based FAQ 4
Nexus IP Console Position – complete communications solution 5
C-Soft – software-based radio dispatch console 6
Advanced SIP – session initiation protocol (SIP) 8
Beacon Series – includes Advanced Digital Headset Box (ADHB-4) and Remote Headset Box (RHB) 9
IP-223 – dual IP network remote adapter panel 10
MTRBi - radio interface for C-Soft 12
Accessories – includes TDI, NI223+, IP-25300, NEO-10 and HB-3+ 13
Network Recorder
14
Remote Database Reviewer
Telex System Manager (TSM)
15
V.I.P.E.R. – IP-based radio control system
IP-1616 – eight-line IP-based radio dispatch console 16
C-6200 – 18-line IP/analog radio dispatch console
17
IP-2002 – two-line IP-based radio dispatch console
C-1616 – six-line analog tone remote control console 18
C-2002 – two-line radio control console
19
C-2000 & C-2000HS – single-line radio control console
DSP-223 & TRA-223– tone remote adapter panel
20
Headsets – includes DH2000, DH2200, DH3000 and DH3200
Microphones – includes MDMS, 6513C, DT-GN-18 and PC desktop-18RD 21
System Diagram 22
2 | www.telex.com/radiodispatch
3. Telex Radio Dispatch, part of the Bosch Group, manufactures and delivers
thousands of mission-critical communication systems worldwide.
Telex Radio Dispatch is the leading manufacturer of IP screen to your application. Standard features include Instant
control for two-way radio communications. Based on a Recall Recorder, paging, and intercom. Available features
distributive architecture, Telex dispatch console systems include Fleetsync, MDC-1200, and Advanced SIP. C-Soft is
have flexibility, scalability, and redundant capability based compatible with Windows XP and WIN7 “32 bit” formats.
on the network.
The newest addition to the list of Radio interfaces is
Telex converts audio and control functions from analog or the connection that gives the C-Soft consoles control
digital to Ethernet packets. Once converted to IP, the signal capability of MOTOTRBO™ Professional Digital Two-Way
can be transported via LAN, WAN, 802.11 wireless, satellite, Radio Systems. All of the MOTOTRBO™ Conventional and
and the Internet. With this many mediums to work with, Capacity Plus features are supported at this time, including
systems can be precisely scaled according to application, GPS tracking. Add this capability to the most scalable and
whether confined to a single building or campus, or flexible IP-based dispatch control system, and you have the
covering an entire country or the world. You can control a best dispatch solution offered today.
Telex IP based system in Texas from New York, and all with
parallel control in London, England. Another key component is technical support and services.
Telex Technical Support team members have the right
C-Soft is the industry’s most flexible and capable dispatch combination of extensive training and years of experience
software – the perfect application for any dispatch to assist you with any technical issues, ensuring your Telex
environment. This software installs on a Telex Nexus equipment provides the right solution for you.
position, a Telex laptop, or on your own computer position
(when used with the ADHB-4 audio adaptor). With two to Read on to learn more about how a Telex dispatch solution
two hundred radio lines, C-Soft allows you to design your can be right for you.
“ Telex for Air Logistics LLC
With four dispatch centers and 33 remote sites spanning
a vast triangle from Brownsville, TX to Mobile, AL and
out into the Gulf of Mexico, Air Logistics LLC (a Bristow
Company) of New Iberia, LA, is a leading provider of
helicopter transportation services to the oil and gas
LA, two in Patterson, LA, two in Intracoastal City, LA,
and two in Galveston, TX. We can control our radios at
any of these locations, with full redundancy built-in. If we
lose connectivity in New Iberia, we can have a dispatcher
in Patterson or Galveston pick up the communications
industry. Air Logistics relies upon a digital network of Telex seamlessly... The IP-223/C-Soft setup gives us complete
equipment for flight following communications with their flexibility; our operators can operate any radio from any
aircraft. console, which is a very big plus. When an operator goes
on break, we can pair-up different radios together. They
“Our remote stations are equipped with IP-223 VOIP can pick and choose what console controls which radio.
adaptors on an MPLS network,” reports Pete Talbot of Plus, the interface is user-friendly. New operators can be
Air Logistics, “tied back to our C-Soft consoles. We have trained-in in a couple of hours.”
six consoles at our central dispatch facility in New Iberia,
— Pete Talbot
Air Logistics LLC (a Bristow Company)
www.telex.com/radiodispatch | 3
4. IP-based
FAQ
Can we use our existing computer network to create an IP-based
dispatch system, or do we need to build a new one?
The answer to this question depends on the IP dispatch effective as possible. We can work with non-multicasting
system application. In many cases we can use existing IP networks but there are some limitations.
networks, but in other cases, like public safety applications,
• o you want to connect and communicate between multiple
D
you may want to think about creating a secure, standalone
locations or installations via an IP dispatch network? If so,
communications network. Here are a few things to consider
that means you have to have a good network connection
with regard to IP dispatch solutions:
between them. Anywhere you have a network connection
• ow much traffic is on my existing network and
H could be a potential location for communications equipment.
how much bandwidth is available to dedicate to a If you have offices across the country that are all connected
communications solution? via a network, you could communicate between them using
two-way radios because the network ties them together.
• oes your network support multicasting? Multicasting is
D
an important element in making our dispatch solutions as
How much bandwidth will the dispatch system use on the network?
With a C-Soft low-bit vocoder, there can be a bandwidth bandwidth. Multiply that times the number of radios you have
savings of up to 30 percent. This depends on how many on the system—eight radios means 8 x 50 kBit or 400 kBit
radios and dispatch positions you want on the system. The for effective simultaneous communications. Always make sure
breakdown is simple: every device you connect to the system the network has the capacity to account for the maximum
that operates in simplex mode requires 50 kBits of available possible number of simultaneous transmissions.
How important are SIP capabilities in a dispatch system?
The Session Initiation Protocol (SIP) is a signaling protocol important features is the crosspatch capability of radio
used for establishing sessions in an IP network. For a and phone. Now a police officer on a radio channel can be
radio dispatcher, SIP is one way of bringing control of connected to a phone call to coordinate activities or assist
telephone onto the console, be it a simple two-way call or dispatch with a critical situation. SIP can be configured to
a collaborative conference session. This allows dispatch use existing switches, so a large number of lines are available
to control phone calls incoming and outgoing phone calls if required. SIP is becoming an important part of dispatch
and use features such as hold/unhold, call forwarding, consoles. Like the IP console SIP is a powerful, yet simple tool
conferencing, voice mail, and call transfer. One of the most which is also very flexible and scalable.
I have two dispatch locations, but can only afford to update one
console to IP. Will I be able to communicate with the old console?
In most cases, Telex will be able to interface an IP-223 or network, the analog console can be monitered and operated
gateway to each channel at the location where the radios like it was another IP console. The flexibility of this design
would be interfaced to the analog console, usually the Central provides a way for an end user to start a migration path to a
Electronics Bank (CEB). The IP-223 is configured in console complete IP change over when replacing the dispatch cosoles in
mode, and allows the channels configured this way to be put on multiple locations. When required, this concept also provides a
the same IP network system as the new IP console. Through the very cost effective backup console to existing analog consoles.
4 | www.telex.com/radiodispatch
5. Nexus IP Console Position
Complete To build your dispatch position:
Communications Choose a PC platform, monitor, C-Soft, headset adapter,
and accessories.
Solution 1. Choose your Computer
Either purchase our Nexus computer, or purchase a
computer that meets the PC Requirements listed on the
next page under the C-Soft product.
2. Choose your C-Soft license size
3. Choose your monitor:
• 19” LCD monitor
• 17” TS LCD monitor
• 19” TS LCD monitor
4. Choose your headset adapter:
• BEACON ADHB-4
• BEACON RHB-1
• ADHB
5. Choose any of the applicable accessories:
• Microphone
• Speaker
• Headset
• Footswitch
1 2 3 4 5
The Nexus IP console position delivers everything for Flexibility and Scalability
dispatch communications—stability, performance, and The Nexus IP console position can be ordered in
world-class dispatch capability. The IP platform makes it configurations from two to 200 lines. It is our most
simple to install, easy to expand, and flexible enough to capable and highest capacity dispatch solution. The user
use in any dispatch setting. interface is completely customizable, meaning you can
control the button layout – the size, shape, color, and even
Service and Support the labeling. Change the background color, create simple
By standardizing around a single dispatch position or advanced dispatch interfaces — the options are nearly
platform, we have been able to optimize both the limitless with the Nexus console position. You can even
operating system and dispatch software for maximum store multiple dispatch configurations on a single station
stability and performance. And we are able to deliver a for different applications or usage scenarios.
total solution that is significantly enhanced and much
easier to support by removing the variables associated
with software installation on an end-user provided PC.
www.telex.com/radiodispatch | 5
6. C-Soft
Software-based Radio Features:
Dispatch Console Available configurations – available in configurations from 2 to 200 lines.
User interface – user-controlled configurations for any dispatch application.
Signaling capabilities – MDC-1200 encode and decode, MOTOTRBO™,
NexEdge, FleetSync encode and decode, DTMF, serial and OTA FleetSync,
5/6 tone (supports emergency, group, individual and status calls).
Instant recall recorder:
• Tracks the last ten minutes of both select and unselect speaker audio.
• Buttons can be set up to start playback at various points in the buffer, or
played call-by-call from the call buffer.
Information windows – per line call history, active emergency, emergency
history, manual call list, and status window.
Intercom capabilities – intercom communications between dispatch positions
can be set up on all consoles on the system.
DTMF keys – a full, 16-key keyboard is supported.
Paging:
• Multiple paging formats are built into the C-Soft console software.
• Quickcall in both the 100 and 1000 group formats, as well as DTMF,
Knox paging tone, and 5/6 tone paging.
• Manual frequency entry mode is also supported.
PC requirements: Alert tones:
• perating system: Windows XP and
O • Three alert-tone types are supported, including: steady tone, pulsed
tone, and hi-lo warble.
WIN7 “32 bit” formats
• All frequencies and durations are programmable.
• Network connection: 10 Mbps or
100 Mbps TCP/IP connection Programmed group mute buttons – for both functions lines can be
• Processor speed: Intel Pentium dual selectively included within these programmed buttons, allowing for instant
CPU 1.8 GHz access to particular lines of interest.
• Memory: minimum of 2 GB of RAM Crosspatch – up to 30 simultaneous crosspatch groups are supported.
recommended
Status indicators – 24-hour clock, VU meter, PTT indication, and Instant
• Parallel applications: not Recall Recorder progress are displayed on the upper status bar.
recommended to run other
applications on PCs running C-Soft, Flexible audio interface options – using Telex’s new ADHB-4 and RHB,
especially those with high demands. C-Soft can interface with all common dispatch communication audio sources,
including headsets, desktop microphones, external speakers (up to six), and
NOTE: These are minimum requirements and users footswitches.
should bear in mind that when handling a large
number of lines - 50 or more per PC – it is strongly
recommended that more powerful computers
Multiple vocoders – per line vocoder type ability to select lower bandwidth
and more robust network resources be deployed. vocoder.
Please consult your integrator for specific system
recommendations. Special interfaces – MOTOTRBO™ , iDEN, TETRA, P25 and phone/PSTN.
6 | www.telex.com/radiodispatch
7. C-Soft is the industry’s most flexible and capable software assets, and the ability to save multiple configurations
dispatch console — perfect application for any dispatch on a single computer. This proven application has been
environment. C-Soft delivers all of the dispatch capabilities deployed in communication centers around the world, in
you expect, while also giving you the flexibility that only an applications from 911 dispatch to mobile command centers
IP-based software console can provide: simple and quick and transportation management.
deployment in the field, easy back-up of communications
Telex C-Soft: A Proven Life-Saver for Okanogan County 911 Center
As the third largest county in the continental United States, The dispatcher tells the responding unit what they know,
Okanogan County spans 5,281 square miles of rugged and the unit gives the dispatcher status updates as they
north-central Washington terrain that includes parts of respond. All of this back-and-forth with the radios that
the Cascade Range and the Columbia River Basin. With a transmit over the air and talk to the Kenwood field radios is
population of about 40,000 dispersed over such a large routed through C-Soft and the IP-223s.”
area, effective distance communication is critical at times
of emergency. “Our number-one focus is life-safety,” says The C-Soft system supports flexible signal routing and the
Emergency Manager Scott Miller of the Okanogan County tailoring of user interfaces to custom needs. And with its
Sheriff’s Office. “We cannot afford to have our radio Radio over IP (RoIP) architecture, all the traffic that would
systems go down at any time.” previously have required a multitude of analog audio lines
between the dispatcher’s station and the radio transceivers
Having built its emergency communications capabilities can now be handled over a single CAT-5 Ethernet cable.
around a Telex C-Soft platform several years ago, the ROIP also gives the county far greater extensibility to
county’s Emergency Management Department has had remote locations. “We have a control station up on a
plenty of opportunity to determine whether the system mountaintop,” Miller says, “and by putting in a network
meets requisite standards for reliability and performance. switch we were able to extend the network to that location
The verdict? “It’s been solid as a rock for five years,” Miller via microwave, which is far more reliable than leasing a line
says. “We’ve had absolutely no problems.” from a phone provider.”
Headquartered in Okanogan, the county seat, the Sheriff’s At the time of the county’s initial purchase, RoIP was still
Office operates the county’s 911 communications center, relatively new in the radio dispatch field, which meant that
receiving all 911 calls and dispatching all first-responders there wasn’t a long track record to go by when selecting
to in-county emergencies. The facility is equipped with a manufacturer. “Looking back,” Miller says, “it’s now clear
the IP-based C-Soft radio dispatch and signaling console that we made the right choice. We are very, very pleased
as well as 25 IP-223 dual IP remote adapter panels, which with the C-Soft system. The system remains state-of-the art
handle conversion between digital and analog signals. because Telex is continually developing the product with
The system also includes a number of DSP-223 tone- updates and advancements that they have not charged us
remote adapters, as well as two Telex Network Recorders for. Plus, Telex’s support has been outstanding, and we are
which perform real-time call recording, supports remote confident that they will still be here to support us five or
monitoring, and includes a searchable database for storage ten years down the road.”
and retrieval of call information. “We really like the ability
for supervisors to be able to call up recordings from “The bottom line for us,” Miller continues, “is that we use
anywhere in the County IP network,” adds Miller. the C-Soft system in a critical life-safety setting, and we’ve
had five years of running 24/7 without a hitch. That’s quite
“We handle all of our public safety radio communications a testament to the system. We would highly recommend it
via the C-Soft console,” Miller continues; “it’s how we to any other 911 center.”
communicate with all first responders in the county. The
911 call comes in, the dispatcher takes the call, determines
the nature of the problem, and then uses C-Soft to either
communicate directly with a specific first-responder unit,
such as a nearby ambulance, or to send out a general page.
www.telex.com/radiodispatch | 7
8. Advanced SIP
Session Initiation Features:
Protocol (SIP) Call hold – places the current call on hold and returns to the previous call.
Call waiting – sends an audible indicator when a third party calls in.
Blind call transfer – two parties are in a call and one transfers the call to a
third party without first contacting the third party.
Call transfer with consultation – two parties are in a call, then the third party
is contacted to announce the transfer before it happens.
Call-forwarding feature:
• Unconditional – routes all incoming calls to voicemail or another number
for any reason.
• Busy – sends a call to another phone number or voicemail in the event
the line is busy.
• No answer – sends the call to another phone number or voicemail in the
event that there is no answer (after a pre-defined time).
Three-way call conferencing – allows up to three calls via conference bridge.
Cross-patching radio PTT users with SIP calls – allows dispatchers to
interface radios via SIP, to be included in the interoperable conferences.
Call conference up to five users – allows for multiple calls out to others,
putting all calls on hold then bridging the conference call.
Do not disturb – allows all calls to be routed to voicemail.
All SIP call sessions are recorded on the Network Recorder – C-Soft records
via echo packets, so the user will have to configure packets in C-Soft Designer.
We are excited to launch a full industry-standard SIP solution. With the Advanced SIP features, users can
solution as a part of the C-Soft software-based console. implement and configure many telephone lines into
Users can opt for a two-line or six-line HW security key a C-Soft console position without having to connect
or SW option for the ADHB-4. Adding our new Advanced external hardware.
SIP solution to the Telex IP dispatch console allows for a
complete console solution. Our application is easy to install and easy to use! The
icons are intuitive and they indicate various calling
Our new application provides a great alternative method features (i.e. call is on hold, etc).
of telephone line installation into a dispatch console
“ Telex ROIP for Cobalt Equipment’s mobile ECC
Cobalt Equipment and TelePath chose the Telex IP-223
ROIP / C-Soft dispatch system because it is the best
solution for a mobile response unit. “Telex is very user-
friendly, It can be changed quickly, and it’s very flexible.
We looked at several other systems, but we kept on
coming back to Telex. A lot of states and state fire
agencies also use Telex, so we figured that it would be a
good fit with anyone with whom we would ever work.”
”
— Rob Knabe
President, Cobalt Equipment
8 | www.telex.com/radiodispatch
9. Beacon
Series
ADHB-4 RHB
Advanced Digital Features:
Headset Box Ethernet communication LED power and PTT indicator
Six audio channels 12 VDC operation
One dual-channel 1/4” headset jack Two remote headset box connections
One XLR connector to low-
Volume control knob
impedance microphone
One desk mic jack LED for power/PTT indication
One telephone handset jack Programmable gain control
Remote Headset Box One NENA I/O jack with offhook Footswitch inputs for PTT and
detection monitor
Separate headset volume knobs Two relays with form-C contacts
Supports three pairs of speakers AUX inputs are DC isolated
Use with any standard amplified
Color LCD-type display
speaker
The all-new Advanced Digital Headset Box (ADHB-4), an backwards-compatible with Windows XP. In addition,
enhancement to the HB-3+, is our next generation radio the ADHB-4 includes all connections necessary for
dispatch headset adaptor. The ADHB-4 works exclusively full integration into the most common radio dispatch
with the Telex C-Soft console version 5.000 or later. configurations.
It processes audio internally and communicates with
C-Soft to transfer the signal via USB, which eliminates One of the most striking features of the ADHB-4 is its
dependence on the PC sound card. This allows users to
full-color LCD screen. This display hosts a rich interface
purchase their own computer. While the current HB3+
which provides the user with at-a-glance system status
works with two speakers, the ADHB-4 supports up to six
updates. With its Web capability, the ADHB-4 can also be
speakers per position.
managed via web interface. These advanced configuration
The ADHB-4 is the heart of the C-Soft console dispatch options will bring peace of mind to dispatch operators,
position. It removes the barrier between users of different enabling them to focus on the critical task at hand. Our
PC audio platforms and enables them to relay vital, life- new Remote Headset Box (RHB) allows up to two option
saving information. To further enhance the flexibility RHB’s to connect to the ADHB-4. This allows other users
of the system, the ADHB-4 works with most desktop to listen to the dispatch position (ie: supervisor, second
systems running the latest Microsoft Windows 7. It is also dispatcher).
“ Telex ROIP heads for the Stratosphere Casino in Las Vegas
“Our PBX operators need to stay focused on the phones
and providing guest service rather than navigating
screens or using complicated technology,” Lawney adds.
“The interoperability between the phones, the GUI on
the computer screen and the IP-223 systems behind the
channels are arranged so that they can see them clearly
without scrolling; they don’t need to do anything beyond
click on a screen button to view, scan, and send the call
messages out.”
“Working with McIntosh and Telex was a great
scenes is as user-friendly as it is technologically seamless.
experience. There were no issues, no down time, and the
The operators have the usual buttons that you would see
system is a perfect fit for us.”
on a regular radio-base station on their screens and the
— Melissa E. Lawney,
Telecomm Specialist for American Casino Entertainment Properties LLC
www.telex.com/radiodispatch | 9
10. IP-223
Dual IP Network Features:
Remote Adapter Radio telephony operation – allowing local consoles to change to a remote
radio channel via POTS (Plain Old Telephone Service) line.
Panel
Enhanced crosspatch capabilities:
• Line-to-line crosspatch – enable and disable via DTMF strings.
• Start/stop function tone line-to-line crosspatch – designated function
tones have the ability to automatically set up and knock down line-to-
line crosspatches within the device.
• Dial – remote user with portable radio can key a DTMF string, causing
IP-223 to take the TDI off-hook, dial a pre-programmed phone number
and establish a patch between the devices via DTMF strings.
• Dial VoIP – remote user with portable radio can key DTMF string, causing
Available options: an IP-223 to join different multicast group and port-mapping the IP to a
• leetSync encode and decode
F different channel.
• MDC-1200 encode and decode • Phone patch – remote user with portable radio can key a DTMF string,
causing IP-223 to take the TDI off-hook. The user can then manually dial
• iDEN interface with NI-223 a phone number.
• TDI telephone interface required if • Multiple vocoders (per line vocoder type) – ability to select lower
using POTS bandwidth vocoder.
• Multiple Vocoders Kenwood P25 TK5710/5810 serial control – supports encode and decode of
FleetSync ID and P25 ID, channel change, scan ON/OFF, and monitor. Also
capable of direct serial control of Kenwood 80, 90, and 150 series radios.
Generate FleetSync MSK signal at the IP-223 – DMR (Digital Mobile Radio)
radio: NexEdge does not require specific Kenwood base station.
Advanced compatiblity with multiple radio manufacturers – Motorola,
Kenwood, EF Johnson, iDEN, and Sepura.
Five-tone detection – decodes five-tone messages received from remote
radios and sends console information to the display.
COR click dialing – while using COR (Carrier Operated Relay), the remote
user can key click a portable radio, causing IP-223 to take TDI off-hook and
dial a pre-programmed phone number to establish a telephone connection
between remote user and a designated dispatch console.
Improved web-based programming interface – redesigned web page
displays important information on start screen and simplifies navigation to
critical programming areas.
Encode iDEN emergency – able to receive and decode ID and information
related to incoming iDEN emergency signals.
Sepura status messaging – able to decode and display status messages
generated via TETRA (Sepura SRM200/3500) radios.
Telex System Manager (TSM) – easily detects Telex device on the network
for easy firmware upgrade and configuration.
10 | www.telex.com/radiodispatch
11. The Telex IP-223 IP Network Remote Adapter is the center network; that network can be within a single building, or
of our IP solutions. The IP-223 bridges two-way radios and can reach across the entire country—wired or wireless. The
other communications devices onto the IP dispatch network. IP-223 also uses multiple types of communications interface:
Each IP-223 allows you to connect and control up to two IP, two-wire, four-wire, iDEN, local control, and TDI with
communications devices from any dispatch location on the phone.
IP-223 also offers seven functional modes:
• Local – direct connection to any radio, bridging it • Phone – connects a standard POTS telephone line to
onto the IP network. the dispatch network via the TDI.
• Tone – generate standard control tones via • iDEN – puts iDEN phones onto the dispatch network
conventional connections to radio. and provides advanced access and control.
• Console – bridge analog consoles into an IP dispatch • TETRA – provides access to advanced features of the
network. TETRA system via an interface with Sepura radios.
• Crosspatch/repeater – directly patch communications
devices on the network without a console. It can also
be used to extend coverage.
“ Telex ROIP for San Ramon Valley’s mobile ECC
Located in the San Francisco bay area, this one of many
busy emergency service departments relying upon the
solid performance of Telex IP-223 IP-based network
remote adapters for their mobile command vehicles.
don’t have the consoles set up to change channels—we’re
using all the capabilities of the radios. We can travel all
over the bay area providing communications, and are
not restricted to 2, 16, 24 channels—the radios select the
channels. The IP-223 units are at the heart of an absolutely
Suter states: “The system is easy to program and set up;
user-friendly dispatch system.”
we like how it provides channel-to-channel interoperability
with each console. Because we’re using mobile radios, we
”
— Chris Suter
San Ramon Valley Fire District Communications Coordinator
“ Denver Public Schools use Telex for stability and effectiveness
“I have been working with two-way radios for DPS going
on 25 years now, and RoIP is the neatest technology I
have seen in communications yet. The ability to multicast
over Ethernet is a powerful tool. RoIP has created endless
possibilities for our two-way applications. We can design
and add on to the Telex IP-223/C-6200 system in many
different ways. It’s a great platform to grow with.”
”
— Jim Bailey,
Denver Public Schools Radio Room
www.telex.com/radiodispatch | 11
13. Accessories
TDI
Telephone This innovative technology allows dispatchers place and
receive calls from their console. A single analog phone line
Dispatch can now be a shared resource among several IP-based
Interface dispatch consoles in a facility.
NI223 +
iDEN Users are able to change groups, initiate/terminate calls,
crosspatch iDEN calls to other radio channels on the
Interface to network, communicate directly with the phone user, power
the IP-223 the iDEN phone and pass caller ID information back to the
console.
IP-25300
EFJ 25300 Users have the flexibility to crosspatch the radio with any
Interface to other communication platform, including iDEN and a range
of other two-way radios. The EFJohnson 5300 mobile radio
the IP-223 is available in bands from VHF to 700/800 MHz and is used
around the world.
NEO-10
Network Input/Output The NEO-10 is a network-based input/output device that
has 10 DPDT relays and 10 inputs for monitoring external
Control Device events. Anytime a relay or input changes, the NEO-10 sends
a message across the network, allowing all console users
on the system to see status updates in real time. Actual
control of the NEO-10 is accomplished by a TCP/IP socket
connection from the controlling console.
HB-3 +
Headset Adapter Panel Microphone and headset input circuits allow end-users
to choose between electret and dynamic element
microphones. The HB-3+ contains its own microprocessor
and software, giving it the intelligence and ability to
control multiple inputs and outputs. Sold with legacy
hardware consoles.
www.telex.com/radiodispatch | 13
14. Network Recorder
Features:
Faster CPU speed, more RAM Separate OS drive
Dual 500 GB hardware RAID control 250 GB SATA removable hard drive
hard drives for archive
Easy to recover the system in case of Additional archive hard drive
failure available
Exportable
Network Recorder Software
The new network recorder is sold in combinations of 12, 24, 50, and 100 licenses.
The Telex Network Recorder allows you to monitor and The Remote Database Reviewer and Network Recorder’s
record audio for any channel in real time. Stores detailed relationship is as client and server. The Remote Database
information for each call and event in an SQL database Reviewer (client) connects to Network Recorder (server),
for quick and easy retrieval. This includes: Source IP and a session is created. While the session is active, the
addresses, channel changes, crosspatch creation and tear Remote Database Reviewer is able to perform database
down, Supervisor mode start and end, ANI, Date/time/call queries and request audio from the Network Recorder
duration, Line number, Scan status and NEO-10 relay and server. As long as the session is active, the session’s user
input logging. account is also considered active. When the Remote
Database Reviewer is closed, the session ends, and the
The recorder monitors the radio network for audio packets user becomes inactive. The Network Recorder allows for
and records specific criteria. These are stored as raw PCM six sessions at a time.
audio and then compressed into MP3 files. A 32-bit digital
signature is added to the file to guarantee its authenticity. The Telex Remote Database Reviewer software is a state-
Both RX and TX audio are stored and separated for search of-the-art tool that enables users to remotely access
purposes. the Telex Network Recorder’s database of audio files
for playback and data export, to generate a report for
Network Recorder monitor features: portable viewing. The Remote Database Reviewer is able
• Check for heartbeat, warning and errors to stream and copy audio files and data from the Network
Recorder’s archive of recorded audio.
• Established reporting messages –
MP3 compression problems
Database connection/reconnect problems
Protect key (dongle) not found Recorder Search Engine:
Sound card problems The network search engine can search the recorder
Hard drive full computer using these parameters: ANI, line number, date,
Database rebuilding time, and call duration. Unrelated calls can be removed
A line has been recording over 30 min. from the search screen, and calls of interest can be copied
Accumulation of error files for playback on another computer or an MP3 player.
Less than 20 GB left on hard drive recorder Large groups of calls can be archived for permanent
storage and to clear disk space. Archived calls can then
be brought back into the database for later review.
14 | www.telex.com/radiodispatch
15. Telex System Manager
Features:
Option to save the configuration
to a file
FREE!
NO CHARGE Selectively copy device
for this parameters from one configuration
application! to another
Import or export to XML or CSV
file, ID directory, Crosspatch table
Save device configuration files to
local disk for backup, archiving, or
duplication
Record configuration files back to
a Telex device
Requirements:
• Windows XP SP2 or higher
• NET Framework 2.0 or higher
• Windows Installer 3.1
Telex System Manager (TSM) software allows users to the ability to update firmware on the IP-223, IP-2002,
easily configure Telex devices. TSM allows a user to view IP-1616, C-6200, and the NEO-10. Telex System Manager
and manipulate configuration parameters for the IP-223, replaces and improves upon the existing FTP Telex and
IP-2002, and the IP-1616. In addition, TSM includes Configuration Saver programs.
V.I.P.E.R.
IP-based Radio All V.I.P.E.R. packages include:
four Telex IP-223 radio controllers, external connection for up to eight different
Control System portable radios, external Cat-5 network connection, internal network router,
and a rugged weather resistant mil-spec case.
V.I.P.E.R. MCU also includes: V.I.P.E.R. eight also includes:
Nexus IP laptop computer 110-240 V @ 100 W max power supply
12-line C-Soft dispatch console software Closed dimensions:
and network recorder software 28” W x 27.25” D x 15.5” H
Built-in storage drawers for laptop
computer and accessories
110-240 V @ 320 W max power supply
Closed dimensions:
28” W x 29.5” D x 22.5” H
Use the V.I.P.E.R. to create completely self-contained Number of radios needed for control
dispatch networks that are easy to deploy in emergency Each Network Remote Adapter (IP-223) supports up to two
situations, providing effective communications radios. The number of different radio systems you need to
interoperability solutions. V.I.P.E.R. is based on a modular control will determine the number of IP-223s to include.
architecture, and it provides the flexibility to create an exact
Number of pre-configured radios for installation
solution for any application. Any authorized Telex Radio When building a V.I.P.E.R., agencies and integrators often
Dispatch dealer or integrator can help design a system that choose to have their most commonly used radios built right
fits your needs. Our V.I.P.E.R. MCU and V.I.P.E.R. 8 can be into the unit.
connected together to achieve expanded capabilities.
www.telex.com/radiodispatch | 15
16. Hardware
Consoles
IP-1616
Eight-line IP-Based Features: Controls:
Radio Dispatch Simplex/full-duplex operation • onitor, intercom, and PTT button
M
(field programmable)
Console • Up to four alert tones
Cross-mute (Ethernet-based) • Crosspatch
Parallel console update • Group select, two pre-determined
groups
Instant Recall Recorder (IRR)
• Supervisory control
Telex System Manager (TSM) • 16-digit DTMF keypad
easily detects Telex device on the
network for easy firmware upgrade • Volume controls
and configuration. • arallel TX LED detect
P
16-channel control • Channel selection
Cross-patching of two to eight lines • Instant PTT
• Four programmable buttons
Communications with cross-patch
groups while operating on unused • Paging (two-tone, DTMF, manual)
channels • A- and B-menu buttons
Line select call with alarm
The IP-1616 is a workhorse console that offers all the The IP-1616 requires no CEB or additional CPU equipment
dispatch features and control that you would expect from for operation. All processing and control capabilities are
a larger, more expensive solution. Multiple IP-1616s can completely self-contained within the unit. Requires a
be used to control larger operations. Its smaller desktop gooseneck desktop microphone or dispatch headset for
footprint takes up less room at the workstation, but still operation—all sold separately.
offers all the dispatch capabilities and controls you need.
IP-1616 also offers:
• Call history – up to last 50 incoming calls displayed. • Radio telephony operation – allows local console to
• Autodial – dials from history list and phone list. change channel of the remote radio via POTS line.
• Caller ID – displays phone, iDEN, MDC, FleetSync, Also gives operators the ability to designate certain
TETRA and 5-tone. lines to automatically fail-over to a standard POTS
line if the IP connection fails.
• NEO-10 support – two NEO-10 relays from the console.
• Phone line interface – allows interfacing to a phone line.
• iDEN support – full support of NI-223 features,
including ID, Go-ahead beeps, signals and manual dial. • Kenwood P25 TK5710/5810 serial control – supports
encode and decode of FleetSync ID and P25 ID,
• Scan features – scans for supported radios.
channel change, scan on/off, and monitor. Also
• Emergency – decodes incoming emergency signals capable of direct serial control of Kenwood 80, 90,
from supported ANI formats. and 150 series radios.
• Clear/coded transmit – transmits to EFJ RS5300 • Generate FleetSync MSK signal at the IP-223 – does
mobile radio. not require specific Kenwood base station.
16 | www.telex.com/radiodispatch
17. C-6200
18-line IP/Analog Features: Controls:
Radio Dispatch Simplex/full-duplex operation
(field programmable)
• onitor, intercom, and PTT button
M
Console • Up to four alert tones
Crossmute (hardwire) • Crosspatch
Parallel console update • Group select, three pre-determined
groups
Instant Recall Recorder (IRR)
• Supervisory control
Telex System Manager (TSM)
easily detects Telex device on • 16-digit DTMF keypad
the network for easy firmware • Volume controls
upgrade and configuration. • arallel TX LED detect
P
Programmable single or dual • Channel selection
function tones
• Instant PTT
16-frequency control
• Four programmable buttons
Two- or four-wire (field • Paging (two-tone, DTMF, manual)
programmable with optional line
cards), local and EM • Auxiliary, up to four buttons
Programmable squelch control
Paging (multiple formats)
Supports NEO-10 controls
The C-6200 is a unique platform in the dispatch hardware console back-up to the Nexus IP dispatch
industry that can function as either an IP-based or an position. The C-6200 requires no CEB or additional CPU
analog console, giving you the flexibility to deploy it equipment for operation. All the processing and control
in numerous settings. Perfect for any small to mid- capabilities are completely self-contained within the unit.
sized operation, the C-6200 offers world-class dispatch Requires a gooseneck microphone, desktop microphone,
capability and can even be configured to bridge analog or dispatch headset for operation—all sold separately.
and IP assets within a single unit. It’s also the perfect
IP-2002
Two-line IP-Based Features: Controls:
Radio Dispatch Simplex/full-duplex operation • onitor, intercom, and PTT button
M
(field programmable)
Console • Up to two alert tones
Crossmute (hardwire) • Crosspatch
Parallel console update • Supervisory control
Instant Recall Recorder (IRR) • 16-digit DTMF keypad
• Volume controls
Telex System Manager (TSM)
easily detects Telex device on • arallel TX LED detect
P
the network for easy firmware • Frequency selection
upgrade and configuration.
• Menu button for direct menu access
100 talkgroup/frequency control
• Paging (two-tone, DTMF, manual)
Call history with autodial
Caller ID (phone, iDEN, MDC,
FleetSync, TETRA and 5-tone)
Scan feature for supported radios
Emergency – decodes incoming
emergency signals from supported
ANI formats
The perfect footprint for smaller operations or supervisory IP-2002 on the network. The IP-2002 requires no CEB or
monitoring situations, the IP-2002 is an IP-based dispatch additional CPU equipment for operation—all the processing
console in a familiar desktop telephone form factor. and control capabilities are completely self-contained
Dispatchers using the IP-2002 can initiate a cross-patch within the unit. The console comes with a handset and
between the two lines, as well as inject audio into the panel mic. Other microphone options are sold separately.
cross-patch. A simple Ethernet connection places the www.telex.com/radiodispatch | 17
18. C-1616
Six-line Analog Tone Features: Controls:
Remote Control Two- or four-wire per line • elect/Unselect status for each line
S
(field programmable)
Console • Selective call indication
Simplex/full-duplex per line • 16-function tone button selection
(field programmable) • TX all button
Programmable squelch control per • RX all button
line
• Mute button
TX monitor
• Alert button
Supervisory control • UX relay button
A
16-channel control • Intercom
Two alert tone cadencing • PTT button
(keypad programmable)
• 16-digit DTMF keypad
Crossmute per line (hardwire)
• Supervisory button
TX notch filter
• TX detect LED for selected audio
Wildcard groupings
• Line activity monitor LED for each
(function tones)
line
The C-1616 is designed for easy field programmability. Its numeric indication, and features a clock and audio-level
modular design offers selection and control of up to six base meter. Multiple consoles can be easily programmed by
stations and 16 frequencies. The C-1616 comes standard using the serial port located on the back of each console.
with two channels. Additional channels may be added by Unlike other manufacturers’ equipment, the C-1616 requires
installing another two-line module—sold separately. Its no additional programming. Optional: handset/headset,
unique vacuum florescent display provides channel alpha/ gooseneck mic, desk mic, and footswitch.
“ Telex brings flexibility to University of Phoenix Stadium
“The 24-hour stadium security team covers a multitude
of responsibilities during its rotating shifts, including
video surveillance, fire alarms, door/gate monitoring and
answering after-hour incoming phone calls. An easy-to-use
dispatch system was essential, especially in terms of training
new staff and ensuring seamless operator turnover at the
primary dispatch position.”
Creative Communications recommended a Telex C-Soft
12-line basic dispatch console, using IP-223 to interface
remotely with CDM base radios.
”
— Nick Spiro,
Creative Communications
18 | www.telex.com/radiodispatch
19. C-2002
Two-line Radio Features: Controls:
Control Console Selective call indication • onitor, intercom, and PTT button
M
Parallel console update • Alert tone
• ALT button
Alert tone
• Mute, release, and select (per line)
Time duration of the PTT
• Supervisory control
Audio delay
• 16-digit DTMF keypad
Function tones (programmable) • Volume controls
Two- or four-wire (field • arallel TX LED detect
P
programmable), local and EM
• Frequency selection
Simplex/full-duplex • Three simultaneous microphones
(field programmable)
Programmable squelch control
Crossmute (hardwire)
TX monitor
Supervisory control
TX and RX notch filter
Programmable TX delay
Compact, but still loaded with features, the reliable programmed by using the DTMF keypad on the front of
C-2002 offers cross-mute and supervisory capability the console. Used with our mating DSP-223 series adapter
and programmable squelch control, which eliminates the panels, the C-2002 meets all the needs and requirements
unwanted noise generally associated with line monitoring. for controlling remote base stations. The console comes
The C-2002 can control two base stations and select up with a handset and panel mic. Optional: headset, desk mic,
to 99 frequencies. This DSP-designed console can be footswitch, and wall-mount kit.
C-2000 C-2000HS
Single-line Radio Features: Controls:
Control Console Programmable single or • onitor button
M
dual-function tones • Intercom
Two- or four-wire • PTT button
(field programmable) • 16-digit DTMF keypad
Simplex/full-duplex • Volume controls
(field programmable)
• arallel TX LED detect
P
Programmable squelch control • Frequency selection
TX monitor
Supervisory control
Cross-mute (hardwire)
TX notch filter
Alert tone/warble
15 programmable DTMF addresses
Parallel console update
The C-2000 allows dispatchers to select and control a single Unlike other manufacturers’ equipment, the C-2000 requires
base station and up to 100 frequencies. It’s also designed for no additional software. The C-2000 console comes with a
easy field programmability using the DTMF keypad. Used built-in mic. Optional: headset and desk mic. The C-2000HS
with Telex’s DSP-223 series adapter panels, this console includes handset. Optional: desk mic and footswitch.
meets all dispatchers’ needs and requirements for controlling
remote base stations. Multiple consoles can be programmed
by using the serial port located on the back of each console.
www.telex.com/radiodispatch | 19
20. DSP-223
Tone Remote The Telex DSP-223 provides a reliable means of remotely
controlling two-way-radio base stations. The adapter can
Adapter Panel be used in conjunction with all radio dispatch consoles, or
other manufacturers’ (such as Motorola and GE) remote
consoles that use the industry-standard sequential tone-
keying format. The DSP-223 is interconnected to the
distant remote control console(s) by any voice-grade
transmission medium, such as a microwave link, leased
telephone line, or a twisted-pair 600-ohm line. All DSP-
223s are capable of decoding the PTT (push-to-talk/
transmitter-on) tone sequence and the voice-plus-tone
signals during transmission. All models are prepared for
jumper plug conversion from two-wire line operation to
four-wire line operation. In the four-wire mode, the panels
are full-duplex capable.
TRA-223
Tone Remote The TRA-223 tone-adapter is a simple way to remotely
control radio base stations. The TRA-223 can be used
Adapter Panel in conjuction with all Telex analog consoles, or other
manufacturers’ consoles that use the industry-standard
sequential tone keying format.
Base stations can be connected to the distant remote
control console(s) by any voice-grade transmission
medium-microwave link, a leased telephone line, or a
twisted pair 600 ohm line, and are capable of decoding
the PTT and monitor function tone sequence. The TRA-
223 also features a front panel dip switch that allows you
to select various options, such as two- or four-wire line
operation, along with full-duplex.
DH2000
Single-side Headset Features: The DH2000 DH2200 headsets
leverage technology from the Telex
Weighs 3.2 ounces
Airman 750 headset, which is a best-
Mic-noise-canceling electret in-class lightweight headset. A flexible
DH2200 microphone
Ear seals foam
boom allows microphone adjustment
to preferred side, and the adjustable
stainless steel headband makes it
Dual-side Headset Three-year warranty comfortable—even on the longest of
shifts.
DH3000
DH-3000 DH3200 are our noise-
Single-side Noise- Features: canceling headset options, which are
Canceling Headset Weighs 4 ounces very light, with ear cups and cushions
Total noise reduction 12 dBs enlarged from previous models to
enhance comfort. Other features
DH3200 Volume control
Mic-noise-canceling electret
include a set-and-forget volume
control and fully flexible boom. This
microphone headset dramatically improves the
Dual-side Noise- clarity of communication and does not
Ear seals plush foam
Canceling Headset require batteries or panel be power-
Three-year warranty active noise reduction powered by
microphone bias.
20 | www.telex.com/radiodispatch
21. MD-MS 0118022
Omni-Directional Specifications:
Electret Microphone Type: Dynamic
Directivity: Omni-directional
Sensitivity: -14 ± 4 dB at 1 kHz (0 dB=1 Vmicrobar)
Frequency Response: 200 Hz-5 kHz
Cable: 4 conductor, 2 shield, 1.5 m 3 5 cm
Dimensions: H 1.43 mm, W 67.5 mm, L 12.9 mm
6513C 30190500
Noise-Canceling Specifications:
Dynamic Microphone Frequency Response: 125 Hz-5000 Hz Net weight: 822 g (1 lb 13 oz)
Polar Pattern: Cardioid, noise- Switch: Leaf, DPDT, switches external
canceling circuit and shorts/opens mike in off
position
Impedance: 150 ohms Finish: Black
Output Level: -57 dB (0 dB=1 mW/10 Cable: 2.13 m (7ft) long, 5-conductor,
dynes/cm2) 2-shielded, vinyl jacket, black
Case Material: Pressure-cast zinc and Dimensions:
Cycolac 246.1 mm H, 114.3 mm W, 122.2 mm L
DT-GN-18
Desktop Gooseneck Specifications: **windscreen furnished
Microphone Frequency Response: 100 Hz-15000 Hz Color: Non-reflecting black
Generation Element: Condenser, Power Requirements: 1.5 VDC to 9
back-electret VDC phantom supply
Sensitivity, open circuit voltage: 8.0 Output Impedance: Compatible with
mV (-42 dB)/pascal @ 1 kHz RTS keypanels
Power Level, 1 kHz (0 dB=1 mW/ Current Consumption: 500 µA
pascal): -44 dB
Polar Pattern: Cardioid Maximum Head Diameter: 14 mm
Dynamic Range: 102 dB Gooseneck Diameter: 6.4 mm
Mounting: Male threaded TRS Electronics Module Diameter: 20 mm
PC desktop-18RD
Polar Choice 18” Specifications: **windscreen furnished
Microphone Frequency Response: 50 Hz-25000 Hz Output Impedance, 1 kHz: 200 ohms
Generation Element: Dual condenser, Equivalent Noise: 26 dB SPL “A”
back-electret weighted (0 dB=20 micropascals)
Polar Patterns: Omni-directional, Polarity: Pin 2 positive, referenced to
Cardioid, Super-cardioid and Hyper- pin 3, with positive pressure on the
cardioid diaphragm
Switches Controls: Top mounted Current Consumption: 8 mA with
push-button configuration switches P12 supply
Sensitivity, open circuit voltage, 1 Cable: 10 ft, 5-conductor cable,
kHz: 5.6 mv/pascal terminated with 3-pin male XLR
Clipping Level (1% THD): 135 dB SPL Power Requirements: 12-52 VDC
Dynamic Range: 109 dB Base Dimensions:
175 mm L, 117 mm W, 56 mm H
www.telex.com/radiodispatch | 21
22. System
Diagram
Since Telex IP dispatch utilizes an Ethernet data network as its backbone
for all the components, communications is possible to and from
anywhere a user has access to an IP network. VoIP networks are easily
expanded to grow with users’ needs, and the number of end users that
can be added to the network is virtually unlimited. Get in touch with
Telex and let us help you control your communicaitons with IP.
Network
Recorder
10101010101010101010101010101010101010101010101010101010101010101
10101010101010101010101010101010101010101010101010101010101010101
01010101010101010101010101010101010101010
IP-2002 10101010101010101010101010101010101010101
01010101010101010101010101010101010101010
10101010101010101010101010101010101010101
01010101010101010101010101010101010101010101010101010101010101010
10101010101010101010101010101010101010101010101010101010101010101
01010101010101010101010101010101010101010101010101010101010101010
IP-223
IP-1616
22 | www.telex.com/radiodispatch radio tower
23. NEO-10
Door control, monitor power,
temperatures, and other
console functions
Nexus Position
with ADHB-4
1010101010101010101010101010101010101010101010101010101010101010
10101010101010101010101010101010101010101010101010101010101010101
101010101010101010101010101010101010101010
010101010101010101010101010101010101010101
101010101010101010101010101010101010101010
010101010101010101010101010101010101010101
01010101010101010101010101010101010101010101010101010101010101010
10101010101010101010101010101010101010101010101010101010101010101
01010101010101010101010101010101010101010101010101010101010101010
IP-223/control radio
Base Station
radio tower www.telex.com/radiodispatch | 23