The document provides configuration steps for Verint Ultra Suite to successfully interoperate with Avaya Communication Manager 2.1. It describes testing the Ultra call recording solution, which uses Computer Telephony Integration to extract call details from Avaya Communication Manager. The solution supports active station-side recording via E1 trunks and Avaya Communication Manager API, as well as passive tapping of analog stations, E1 trunks, and VoIP. [END SUMMARY]
1) The document describes a configuration of H.323 signaling and IP trunks between Avaya Communication Manager 4.0 and Cisco Unified CallManager 5.1.3.
2) An H.323 signaling group and IP trunk group are configured on the Avaya S8710 server to connect to the Cisco Unified CallManager.
3) IP-IP direct audio calling is verified between Avaya IP phones and Cisco IP phones controlled by the Cisco Unified CallManager.
The document summarizes the VISION family of PBX systems, including the VISION Standard 308S and VISION Premium 309P models. It describes the key technical specifications, features, and target customers of the systems. The VISION offers features such as auto-attendant, allowed and denied lists, alternate number dialing, and least cost routing. It is aimed at small and medium businesses seeking an affordable PBX solution.
The document provides an overview of new commands, modified commands, and deprecated commands in the ArubaOS 6.4 Command-Line Interface. It also describes how to connect to the controller using the serial port or Telnet/SSH, navigate between different command modes, and get help with commands.
The document describes testing of the Amtelco RED ALERT emergency notification solution's ability to interoperate with Avaya Communication Manager using ISDN PRI. The testing focused on RED ALERT's ability to initiate and terminate calls via the ISDN PRI trunk, and properly classify call outcomes. The results showed that RED ALERT successfully generated alerts to Avaya Communication Manager and classified call outcomes properly, including recovering from adverse conditions during alert generation.
The document presents requirements for a Service Control Point (SCP) project for Level 3 Communications. It includes:
1. Executive summary outlining the project objectives to design a consolidated SCP (C-SCP) to integrate existing services, allow for new service creation and changes, and communicate with Level 3 network elements.
2. Sections on customer business requirements, user requirements, and system requirements which specify capabilities for the C-SCP such as supporting a high volume of calls, redundancies, monitoring, reporting, and software upgrades with no degradation of service.
3. The requirements cover areas like hardware specifications, failover reliability, processing capacity, protocol support, configuration, and power needs.
This document provides a summary of Karthik Venky's professional experience and qualifications. He has nearly 8 years of experience in testing VOIP products using various automation and manual testing tools. Currently working as a Technical Lead at Alcatel Lucent India testing their GPON and VOIP products. Experienced in testing protocols like SIP, H.248 and designing automation frameworks using TCL/TK scripting. Well-versed in IMS networks, VOIP technologies, and industry standards. Provides testing and technical expertise across various domains including GPON, IMS, softphone applications and more.
The document provides an overview of the OmniPCX Enterprise telephone system, which includes:
1. A Call Server that acts as the control center and connects to various hardware components over an IP network.
2. Media Gateways that connect standard telephone equipment like analog phones and support wireless devices.
3. IP phones and multimedia PCs that connect directly to the IP network.
Permaconn was established in 1999 and specializes in GPRS hardware and networks for back to base communication equipment. It has established networks in several countries including South Africa. GPRS is a packet-based radio service that enables "Always ON" connections and reporting within GSM networks. A Permaconn outstation interfaces alarm panels to GPRS networks to send alarm events to central stations. Permaconn offers a range of GPRS and IP products for reliable alarm monitoring networks.
1) The document describes a configuration of H.323 signaling and IP trunks between Avaya Communication Manager 4.0 and Cisco Unified CallManager 5.1.3.
2) An H.323 signaling group and IP trunk group are configured on the Avaya S8710 server to connect to the Cisco Unified CallManager.
3) IP-IP direct audio calling is verified between Avaya IP phones and Cisco IP phones controlled by the Cisco Unified CallManager.
The document summarizes the VISION family of PBX systems, including the VISION Standard 308S and VISION Premium 309P models. It describes the key technical specifications, features, and target customers of the systems. The VISION offers features such as auto-attendant, allowed and denied lists, alternate number dialing, and least cost routing. It is aimed at small and medium businesses seeking an affordable PBX solution.
The document provides an overview of new commands, modified commands, and deprecated commands in the ArubaOS 6.4 Command-Line Interface. It also describes how to connect to the controller using the serial port or Telnet/SSH, navigate between different command modes, and get help with commands.
The document describes testing of the Amtelco RED ALERT emergency notification solution's ability to interoperate with Avaya Communication Manager using ISDN PRI. The testing focused on RED ALERT's ability to initiate and terminate calls via the ISDN PRI trunk, and properly classify call outcomes. The results showed that RED ALERT successfully generated alerts to Avaya Communication Manager and classified call outcomes properly, including recovering from adverse conditions during alert generation.
The document presents requirements for a Service Control Point (SCP) project for Level 3 Communications. It includes:
1. Executive summary outlining the project objectives to design a consolidated SCP (C-SCP) to integrate existing services, allow for new service creation and changes, and communicate with Level 3 network elements.
2. Sections on customer business requirements, user requirements, and system requirements which specify capabilities for the C-SCP such as supporting a high volume of calls, redundancies, monitoring, reporting, and software upgrades with no degradation of service.
3. The requirements cover areas like hardware specifications, failover reliability, processing capacity, protocol support, configuration, and power needs.
This document provides a summary of Karthik Venky's professional experience and qualifications. He has nearly 8 years of experience in testing VOIP products using various automation and manual testing tools. Currently working as a Technical Lead at Alcatel Lucent India testing their GPON and VOIP products. Experienced in testing protocols like SIP, H.248 and designing automation frameworks using TCL/TK scripting. Well-versed in IMS networks, VOIP technologies, and industry standards. Provides testing and technical expertise across various domains including GPON, IMS, softphone applications and more.
The document provides an overview of the OmniPCX Enterprise telephone system, which includes:
1. A Call Server that acts as the control center and connects to various hardware components over an IP network.
2. Media Gateways that connect standard telephone equipment like analog phones and support wireless devices.
3. IP phones and multimedia PCs that connect directly to the IP network.
Permaconn was established in 1999 and specializes in GPRS hardware and networks for back to base communication equipment. It has established networks in several countries including South Africa. GPRS is a packet-based radio service that enables "Always ON" connections and reporting within GSM networks. A Permaconn outstation interfaces alarm panels to GPRS networks to send alarm events to central stations. Permaconn offers a range of GPRS and IP products for reliable alarm monitoring networks.
This release of AirWave includes the following new features and updates:
1) It provides instant configuration of Aruba Instant devices directly through the AirWave interface. It also supports zero-touch provisioning of Mobility Access Switches using Aruba Activate.
2) All charts have been updated to use Highcharts, allowing viewing on mobile devices. New options have been added for customizing charts.
3) A new firewall visibility dashboard allows viewing mobile app usage and performance trends across a network.
4) Support for Adaptive Radio Management version 3.0 is included, providing client health information and matching event details to optimize wireless networks.
The document provides information about the Cisco ASR 5500 chassis and hardware components. The ASR 5500 is a 21RU rack-mount chassis that uses rear cards for I/O and processing and front cards for fabric and storage. It can support a variety of card types, including Management I/O cards, Data Processing cards, Fabric and Storage cards, and System Status cards. The chassis provides redundancy and high throughput connectivity between cards using an internal midplane.
volte call flow - SIP IMS Call Flow - MO and MT Call - Volte Mobile originati...Vikas Shokeen
This document discusses the call flow process for a VoLTE call between two parties (A and B) using an LTE network. It involves the following key steps:
1. Party A's IMS network sends an SIP INVITE message to Party B's IMS network with an SDP offer to initiate the call.
2. Resources are reserved on the LTE networks for both parties. SDP negotiations take place to agree on a codec.
3. Once resources are reserved and preconditions met, Party B's phone will ring. When answered, Party B sends a SIP 200 OK message to complete the call setup.
4. The media path is then established between the two parties
The document provides an overview of new features in the ArubaOS 3.1 software release. Key additions and changes include: AP names and groups to better organize and profile APs; profiles to abstract and apply configurations; a new voice services license adding voice-specific features; FQLN to map APs to locations; firewall rules based on traffic type; per-SSID bandwidth contracts; and enhancements to authentication methods like captive portal, VPN, and 802.1x. The release aims to provide more flexibility in configuring and applying wireless services through its new profiling and grouping capabilities.
This document provides instructions for configuring a Panasonic KX-TDE100 IP-PBX to integrate with SIP trunking services using an Edgewater Networks EdgeMarc E-SBC. The configuration includes setting the PBX's IP and gateway settings, enabling SIP registration and codecs, configuring SIP trunk parameters and authentication, assigning DIDs and caller IDs to extensions, and setting dialing patterns to access the SIP trunks. Completing these steps will allow the PBX to place and receive calls using the SIP trunking service.
The Polnet ACP is an automatic call processor that allows up to nine telephony devices to share a single telephone line, eliminating extra phone lines and reducing costs by up to $400 per month. It automatically routes incoming calls to the appropriate device (phone, fax, modem) and provides security features like access codes to prevent unauthorized access to connected devices. In addition to cost savings, the ACP provides benefits like securing dial-up modems, automatic fax detection, and priority call handling for critical outbound calls.
volte ims network architecture tutorial - Explained Vikas Shokeen
I have described VoLTE IMS Architecture in simplified way . Are you also finding 3GPP Specs complicated & Complex for VoLTE IMS . It covers Role played by individual Networks Elements as mentioned below :-
# VoLTE SIP Handset : SIP Support , UAC , UAS , User Agent , SIP-UA
# Underlying LTE Network : MME , SGW , PGW , PCRF , HSS , Dedicated Bearer , QCI , Default Bearer
# IMS Core : SIP Servers , P-CSCF , I-CSCF , S-CSCF , TAS , MMTEL , BGw , MRF , ATCF , ATGW , IBCF , MGCF , IM-MGW , TrGW
# Voice Core or PSTN Network for Break-in or Break-out Calls
This document discusses IMS ENUM and DNS mechanisms for mapping telephone numbers and SIP URLs. It contains the following information:
1. ENUM is defined as the E.164 Number Mapping that provides a system to unify telephone numbers with Internet addressing by mapping E.164 numbers to URIs like SIP.
2. When a UE invites another party using a SIP URL, DNS is used to resolve the URL to an IP address. But for TEL URLs, DNS cannot resolve it so ENUM is used to map the TEL URL to a SIP URL which can then be resolved.
3. If ENUM query for a TEL URL succeeds, the TEL URL is mapped to a SIP URL which
VoLTE Basic callflows in IMS network v2 - includes Registration, Basic VoLTE Call, SDP, Interconnect, Roaming, highlights important SIP headers for session routing and user identities.
Aftek provides services for verticals such as Telecom, Home Automation, Security Control, Transportation, Energy and Automotive.
We provide business solutions for Mobile and Wireless applications, Embedded systems, e-Business, Real-time applications, Enterprise applications and Networking.
cFrame is an open source automated platform for mobile network performance testing in both real and simulated RF environments. It provides distributed test bed automation allowing for reuse of existing hardware and software resources. The document outlines cFrame's features, test configurations, integration with tools like iPerf, and provides examples of automated test scripts and sample test plans.
The document discusses enterprise E911 capabilities and solutions. It explains that IP telephony systems now allow for more complex environments with multiple buildings and locations, requiring E911 applications to properly route 911 calls and provide accurate location information to dispatch centers. These applications allow grouping of users to share location records and emergency response locations. They also provide benefits like automatic MAC address management and real-time 911 call notifications. The document reviews E911 capabilities and tools available in various IP PBX platforms like Cisco, Avaya, Nortel, and Mitel.
The document tests and summarizes the CableWorld CW-4412 MPEG-4 encoder. It has two independent encoders that can convert analog and digital input signals into MPEG-4 format. It has ASI and IP outputs and can be controlled via Windows software or a web browser interface. Status LEDs clearly indicate the operating mode and any errors of each encoder.
VoIP allows for transmitting voice calls over TCP/IP networks instead of traditional circuit-switched networks. It started gaining popularity in the mid-1990s but had drawbacks due to lack of broadband. VoIP offers unlimited distance, lower costs, and uses standards-based protocols like H.323, SIP, and MGCP. Tadiran deployed VoIP across multiple sites globally using Universal Gateways and IP phones.
High Speed Data Connectivity: More Than Hardware (Design Conference 2013)Analog Devices, Inc.
In wireless communications and data acquisition systems, there is more to consider when designing and implementing a complete solution beyond simply physically connecting a high speed analog module to an FPGA platform. Available hardware description language (HDL) components and software are critical to establish an interface, which is necessary for practical system integration. This session starts with a top-level overview of various physical interfaces that are typically used and provides an in-depth focus on high speed serial JESD204B. Prototype HDL used for these types of boards is covered, along with the specific board components and how they are used to interface to high speed ADCs and DACs. Linux device drivers for the HDL components as well as for the ADI components are presented. This includes a short introduction into the Industrial I/O (IIO) framework, the benefits it offers, and how it can be used in end designs.
SPARSH VP248 is a high-definition VoIP phone built with superior acoustics and elegant design to provide unsurpassed audio quality and rich user experience.
Based on open-standard SIP protocol, SPARSH VP248 is interoperable with any standard SIP infrastructure such as IP-PBX, SIP Proxies, Softswitches and Stand-alone applications.
SPARSH VP248 is designed for power users, knowledge workers and managers for quick access totheadvance system features and functions. A feature-packed IP phoneenables user to work efficiently with advance call handling capabilities
The document proposes an architecture for establishing a distributed IP-PBX communication system using multiple voice registers on different platforms and integrating both packet-switched and circuit-switched networks. It provides background on telecommunication technologies and protocols as well as an example case study of implementing the proposed architecture for a nationwide organization with distributed regional offices connected over an IP network. The case study demonstrates configuration of an Asterisk server and Cisco routers to enable voice communication between the regional branches using both the IP network and public switched telephone network.
This document contains a 50 question assessment on Avaya's Unified Communications solutions. It tests knowledge on topics like Cisco vs Avaya availability and reliability options, appropriate server and gateway selections for different sized deployments, Session Manager and Communication Manager components, features to upsell, and characteristics of network regions. The questions cover a wide range of Avaya solutions including definitions, modular messaging, one-X portal, one-X communicator, and Aura.
- The document provides a summary of a telecom test engineer's experience, qualifications and responsibilities. It includes over 5 years of experience in testing telecom protocols like VOIP, SIP, SS7 and working with tools like Wireshark. Recent experience includes working as a senior telecom test engineer at Polycom testing their SIP endpoints and working on the Microsoft Lync platform.
Data conversion for data acquisition is a two-part process that involves sampling and then converting signals into digital venues. These processes inherently remove part of the complete analog signal in exchange for the power and robustness of digital signal handling. This becomes especially difficult when trying to capture signals at the limits of the resolution and speed of our systems. In this session, learn how to design a data conversion system that minimizes the signal loss to match the signal handling requirements … even on the hard ones.
These Application Notes describe the configuration steps required for Amtelco Infinity Version 5.50.05 to successfully interoperate with Avaya Communication Manager 5.0 using PRI QSIG.
Information in these Application Notes was obtained through compliance testing and additional technical discussions. Testing was conducted via the DevConnect Program at the Avaya Solution and Interoperability Test Lab.
SVS; Reviewed:
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.
This release of AirWave includes the following new features and updates:
1) It provides instant configuration of Aruba Instant devices directly through the AirWave interface. It also supports zero-touch provisioning of Mobility Access Switches using Aruba Activate.
2) All charts have been updated to use Highcharts, allowing viewing on mobile devices. New options have been added for customizing charts.
3) A new firewall visibility dashboard allows viewing mobile app usage and performance trends across a network.
4) Support for Adaptive Radio Management version 3.0 is included, providing client health information and matching event details to optimize wireless networks.
The document provides information about the Cisco ASR 5500 chassis and hardware components. The ASR 5500 is a 21RU rack-mount chassis that uses rear cards for I/O and processing and front cards for fabric and storage. It can support a variety of card types, including Management I/O cards, Data Processing cards, Fabric and Storage cards, and System Status cards. The chassis provides redundancy and high throughput connectivity between cards using an internal midplane.
volte call flow - SIP IMS Call Flow - MO and MT Call - Volte Mobile originati...Vikas Shokeen
This document discusses the call flow process for a VoLTE call between two parties (A and B) using an LTE network. It involves the following key steps:
1. Party A's IMS network sends an SIP INVITE message to Party B's IMS network with an SDP offer to initiate the call.
2. Resources are reserved on the LTE networks for both parties. SDP negotiations take place to agree on a codec.
3. Once resources are reserved and preconditions met, Party B's phone will ring. When answered, Party B sends a SIP 200 OK message to complete the call setup.
4. The media path is then established between the two parties
The document provides an overview of new features in the ArubaOS 3.1 software release. Key additions and changes include: AP names and groups to better organize and profile APs; profiles to abstract and apply configurations; a new voice services license adding voice-specific features; FQLN to map APs to locations; firewall rules based on traffic type; per-SSID bandwidth contracts; and enhancements to authentication methods like captive portal, VPN, and 802.1x. The release aims to provide more flexibility in configuring and applying wireless services through its new profiling and grouping capabilities.
This document provides instructions for configuring a Panasonic KX-TDE100 IP-PBX to integrate with SIP trunking services using an Edgewater Networks EdgeMarc E-SBC. The configuration includes setting the PBX's IP and gateway settings, enabling SIP registration and codecs, configuring SIP trunk parameters and authentication, assigning DIDs and caller IDs to extensions, and setting dialing patterns to access the SIP trunks. Completing these steps will allow the PBX to place and receive calls using the SIP trunking service.
The Polnet ACP is an automatic call processor that allows up to nine telephony devices to share a single telephone line, eliminating extra phone lines and reducing costs by up to $400 per month. It automatically routes incoming calls to the appropriate device (phone, fax, modem) and provides security features like access codes to prevent unauthorized access to connected devices. In addition to cost savings, the ACP provides benefits like securing dial-up modems, automatic fax detection, and priority call handling for critical outbound calls.
volte ims network architecture tutorial - Explained Vikas Shokeen
I have described VoLTE IMS Architecture in simplified way . Are you also finding 3GPP Specs complicated & Complex for VoLTE IMS . It covers Role played by individual Networks Elements as mentioned below :-
# VoLTE SIP Handset : SIP Support , UAC , UAS , User Agent , SIP-UA
# Underlying LTE Network : MME , SGW , PGW , PCRF , HSS , Dedicated Bearer , QCI , Default Bearer
# IMS Core : SIP Servers , P-CSCF , I-CSCF , S-CSCF , TAS , MMTEL , BGw , MRF , ATCF , ATGW , IBCF , MGCF , IM-MGW , TrGW
# Voice Core or PSTN Network for Break-in or Break-out Calls
This document discusses IMS ENUM and DNS mechanisms for mapping telephone numbers and SIP URLs. It contains the following information:
1. ENUM is defined as the E.164 Number Mapping that provides a system to unify telephone numbers with Internet addressing by mapping E.164 numbers to URIs like SIP.
2. When a UE invites another party using a SIP URL, DNS is used to resolve the URL to an IP address. But for TEL URLs, DNS cannot resolve it so ENUM is used to map the TEL URL to a SIP URL which can then be resolved.
3. If ENUM query for a TEL URL succeeds, the TEL URL is mapped to a SIP URL which
VoLTE Basic callflows in IMS network v2 - includes Registration, Basic VoLTE Call, SDP, Interconnect, Roaming, highlights important SIP headers for session routing and user identities.
Aftek provides services for verticals such as Telecom, Home Automation, Security Control, Transportation, Energy and Automotive.
We provide business solutions for Mobile and Wireless applications, Embedded systems, e-Business, Real-time applications, Enterprise applications and Networking.
cFrame is an open source automated platform for mobile network performance testing in both real and simulated RF environments. It provides distributed test bed automation allowing for reuse of existing hardware and software resources. The document outlines cFrame's features, test configurations, integration with tools like iPerf, and provides examples of automated test scripts and sample test plans.
The document discusses enterprise E911 capabilities and solutions. It explains that IP telephony systems now allow for more complex environments with multiple buildings and locations, requiring E911 applications to properly route 911 calls and provide accurate location information to dispatch centers. These applications allow grouping of users to share location records and emergency response locations. They also provide benefits like automatic MAC address management and real-time 911 call notifications. The document reviews E911 capabilities and tools available in various IP PBX platforms like Cisco, Avaya, Nortel, and Mitel.
The document tests and summarizes the CableWorld CW-4412 MPEG-4 encoder. It has two independent encoders that can convert analog and digital input signals into MPEG-4 format. It has ASI and IP outputs and can be controlled via Windows software or a web browser interface. Status LEDs clearly indicate the operating mode and any errors of each encoder.
VoIP allows for transmitting voice calls over TCP/IP networks instead of traditional circuit-switched networks. It started gaining popularity in the mid-1990s but had drawbacks due to lack of broadband. VoIP offers unlimited distance, lower costs, and uses standards-based protocols like H.323, SIP, and MGCP. Tadiran deployed VoIP across multiple sites globally using Universal Gateways and IP phones.
High Speed Data Connectivity: More Than Hardware (Design Conference 2013)Analog Devices, Inc.
In wireless communications and data acquisition systems, there is more to consider when designing and implementing a complete solution beyond simply physically connecting a high speed analog module to an FPGA platform. Available hardware description language (HDL) components and software are critical to establish an interface, which is necessary for practical system integration. This session starts with a top-level overview of various physical interfaces that are typically used and provides an in-depth focus on high speed serial JESD204B. Prototype HDL used for these types of boards is covered, along with the specific board components and how they are used to interface to high speed ADCs and DACs. Linux device drivers for the HDL components as well as for the ADI components are presented. This includes a short introduction into the Industrial I/O (IIO) framework, the benefits it offers, and how it can be used in end designs.
SPARSH VP248 is a high-definition VoIP phone built with superior acoustics and elegant design to provide unsurpassed audio quality and rich user experience.
Based on open-standard SIP protocol, SPARSH VP248 is interoperable with any standard SIP infrastructure such as IP-PBX, SIP Proxies, Softswitches and Stand-alone applications.
SPARSH VP248 is designed for power users, knowledge workers and managers for quick access totheadvance system features and functions. A feature-packed IP phoneenables user to work efficiently with advance call handling capabilities
The document proposes an architecture for establishing a distributed IP-PBX communication system using multiple voice registers on different platforms and integrating both packet-switched and circuit-switched networks. It provides background on telecommunication technologies and protocols as well as an example case study of implementing the proposed architecture for a nationwide organization with distributed regional offices connected over an IP network. The case study demonstrates configuration of an Asterisk server and Cisco routers to enable voice communication between the regional branches using both the IP network and public switched telephone network.
This document contains a 50 question assessment on Avaya's Unified Communications solutions. It tests knowledge on topics like Cisco vs Avaya availability and reliability options, appropriate server and gateway selections for different sized deployments, Session Manager and Communication Manager components, features to upsell, and characteristics of network regions. The questions cover a wide range of Avaya solutions including definitions, modular messaging, one-X portal, one-X communicator, and Aura.
- The document provides a summary of a telecom test engineer's experience, qualifications and responsibilities. It includes over 5 years of experience in testing telecom protocols like VOIP, SIP, SS7 and working with tools like Wireshark. Recent experience includes working as a senior telecom test engineer at Polycom testing their SIP endpoints and working on the Microsoft Lync platform.
Data conversion for data acquisition is a two-part process that involves sampling and then converting signals into digital venues. These processes inherently remove part of the complete analog signal in exchange for the power and robustness of digital signal handling. This becomes especially difficult when trying to capture signals at the limits of the resolution and speed of our systems. In this session, learn how to design a data conversion system that minimizes the signal loss to match the signal handling requirements … even on the hard ones.
These Application Notes describe the configuration steps required for Amtelco Infinity Version 5.50.05 to successfully interoperate with Avaya Communication Manager 5.0 using PRI QSIG.
Information in these Application Notes was obtained through compliance testing and additional technical discussions. Testing was conducted via the DevConnect Program at the Avaya Solution and Interoperability Test Lab.
SVS; Reviewed:
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.
Next Generation Campus Switching: Are You ReadyCisco Canada
We will review the latest evolution within the Cisco Catalyst switching product portfolio including the latest Cisco Catalyst 6800 switches and Cisco Instant Access. For more information please visit our website here: http://www.cisco.com/web/CA/index.html
Central controlled IPRAN uses SDN principles to improve upon traditional IPRAN networks by moving major control functions to a central controller. This allows for free service planning, plug-and-play deployment, automatic protocol configuration, and fast troubleshooting from a single controller rather than individual network devices. The central controller has end-to-end visibility and control over forwarding nodes like base stations, cell site gateways, and aggregation gateways. This architecture simplifies operations and maintenance of the network.
This document is the user manual for the PLANET Technology VIP-000/200/400/400FS/400FO Internet Telephony Gateway. It provides an overview of features and specifications, instructions for installation and configuration, and reference for all CLI commands. The manual contains various notices regarding FCC regulations, liability, trademarks, and revision information to ensure proper and safe use of the gateway.
The document provides details on network design and solutions for an Internet data center (IDC). It describes designs for the IDC network infrastructure including the WAN backbone, firewalls, load balancing, caching, LAN backbone, user access networks, and network management systems. It also provides examples of network designs and configurations that can be implemented for hosting customers within the IDC.
Huawei eRAN 7.0 VoLTE feature deep dive_20140515.pptxQasimQadir3
The document describes various enhanced features for VoLTE including TTI bundling, ROHC, semi-persistent scheduling, and delay-based scheduling. It provides details on how these features work, when they are triggered, parameters used, and key performance indicators. The enhanced features aim to improve coverage, capacity, and quality of VoLTE services over LTE networks.
CÔNG TY CỔ PHẦN THẾ GIỚI TỔNG ĐÀI - NHÀ PHÂN PHỐI TỔNG ĐÀI CHUYÊN NGHIỆP
- Lắp đặt hệ thống CCTV chuyên nghiệp
- Lắp đặt hệ thống Camera giám sát.
- Triển khai hệ thống giám sát cho nhà xưởng.
- Cung cấp các giải pháp viễn thông cho doanh nghiệp
Vui lòng liên hệ:
Mr.Khoa : 0968.878.981
Email: khoa.nguyen@thegioitongdai.com.vn
Website: www.thegioitongdai.com.vn
The document discusses Cisco video conferencing gatekeeper design and deployment models. It covers single and multi-zone campus and WAN deployment models. It also covers network design considerations like quality of service, dial plans, call routing, and the Cisco Multimedia Conference Manager which includes the gatekeeper and proxy components.
Implementing Cisco IP Switched Networks (SWITCH 300-115) is a qualifying exam for the Cisco CCNP Routing and Switching and CCDP certifications. The SWITCH 300-115 exam certifies the switching knowledge and skills of successful candidates. They are certified in planning, configuring, and verifying the implementation of complex enterprise switching solutions that use the Cisco Enterprise Campus Architecture.
The document discusses Microsoft Lync Server 2010 Edge architecture and scenarios. It provides an overview of the Edge topology and components, including the Access Edge Server, Web Conferencing Edge Server, and A/V Edge Server. It then summarizes scenarios for external user access, public IM federation, and NAT traversal using STUN, TURN, and ICE to relay media packets between internal and external clients.
Smart line level transmitter sales presentationPavel Buček
This document provides an overview of Honeywell's SmartLine level transmitters and their benefits over competitors. Key points include:
- SmartLine uses advanced guided wave radar technology and a correlation algorithm to more reliably measure level compared to competitors that use simple peak detection.
- A cloud-based application and validation tool allows for online engineering collaboration to expedite transmitter configuration and reduce commissioning time.
- SmartLine offers benefits like universal transmitter wiring, modular design, advanced displays, and Experion system integration that reduce costs and improve usability.
Surf Communication Solutions provides of MoP (Media over Packet) Triple Play (Voice, Video, and Modem/Fax/Data) conversion solutions to communication equipment manufacturers. These solutions are provided in various integration levels: DSP software ; PTMC boards; DSP hardware/software; and PCI boards. http://www.surf-com.com
Foundation Fieldbus - Control in the FieldJim Cahill
Presented by Emerson's Travis Hesketh at the 2011 General Assembly in Mumbai, India on March 9-10.
Download the file to get the full effect of the slide builds.
An experience is a personal and emotional event we remember. Every experience is established based upon pre-determined expectations we conceive and create in our minds. It’s personal, and therefore, remains a moving and evolving target in every scenario. When our experience concludes and the moment has passed, the outcome remains in our memory. Think about what makes you happy when connecting with your own device and then think about what makes you really upset when things are hard, complicated, and slow. If the user has a bad experience in anyone of these areas (simple, fast, and smart), they are likely to leave, share their negative experience, and potentially never return. Users might forget facts or details about their computing environment but they find it difficult to forgot the feeling behind a bad network experience. When something goes wrong with the network or an application, do you always get the blame?
If the number of spine switches were to be merely doubled, the effect of a single switch failure is halved. With 8 spine switches, the effect of a single switch failure only causes a 12% reduction in available bandwidth. So, in modern data centers, people build networks with anywhere from 4 to 32 spine switches. With a leaf-spine network, every server on the network is exactly the same distance away from all other servers – three port hops, to be precise. The benefit of this architecture is that you can just add more spines and leaves as you expand the cluster and you don't have to do any recabling. Intuition Systems will also get more predictable latency between the nodes.
As a trend, disaggregation seems to be most useful for very large companies like Facebook and Google, or cloud providers. The technology does not necessarily have significant implications for small or medium sized businesses. Historically, however, technology has a way of trickling down from the pioneering phases of existing only within large companies with tremendous resources, to becoming more standardized across the board.
Large venues like stadiums or concert halls are challenging environments for Wi-Fi deployments. Most of today’s phones and tablets carry Wi-Fi interfaces. A safe assumption is that at least one device per person in a stadium carry a Wi-Fi interface. Monetizing those Wi-Fi interfaces with real time information of the event in the venue, targeted advertising, internet access, multimedia and social applications can create new revenues to the owner of the venue, if executed properly.
The document describes the DigiFlex® PerformanceTM Servo Drive DZSANTU-040B080. It is a digital servo drive designed to drive brushed and brushless servomotors from a compact form factor suitable for embedded applications. It operates in torque, velocity, or position mode using space vector modulation. It supports connectivity of up to 3 drives to a single controller over an EtherCAT network using ADVANCED Motion Controls' 'DxM' technology. It provides 40A peak current, 20A continuous current, and has inputs/outputs including analog, digital, and high speed capture channels.