The Polycom PathNavigator is an advanced gatekeeper solution that makes IP and ISDN video conferencing easy to use. It provides benefits such as simplified dialing, automated deployment and management, and intelligent network routing. Key features include conference on demand, least cost routing, call forwarding, and alternate routing to increase flexibility and reduce costs. The solution is designed to integrate with other network components and manage complex video networks efficiently.
The document discusses CoreStor, an IP recording solution from Delma that can capture and record IP traffic, including VoIP packets. It describes various methods for capturing IP traffic, such as using span ports, port mirroring, conferencing, or custom gateways. CoreStor is designed to integrate seamlessly into existing systems and provide recording in a single chassis. It supports standard computer hardware and includes replay, administration, and analysis client software.
The document provides an overview of VoIP components, standards, architectures and implementation choices. It discusses key VoIP elements like terminals, packetized voice, soft switches, media servers, gateways, LANs/WANs and standards. It also describes common VoIP architectures for computer-computer, computer-phone, phone-phone communication over the internet. Finally, it outlines VoIP solutions for businesses using VoIP-enabled PBXs, replacing PBXs with softswitches, and hosted PBX solutions.
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.
The document discusses the features of Aethra Telecommunications' integrated access devices and IP PBX system. The devices provide broadband access, voice and fax ports, and advanced services. The IP PBX system offers standard features like call forwarding and transfer, as well as advanced features with the Micro Unified Communications license, including interactive voice response, voice mail, and call logs for billing. The system supports analog, ISDN, DECT and SIP phones and provides a comprehensive set of features to enable IP PBX and unified communications services for small businesses.
This document provides an overview of integrated access devices from Aethra Telecommunications, including broadband access options, voice and data port configurations, operating system features, and advanced application capabilities. Key products highlighted are the BG and SV series, which support ADSL, VDSL, SHDSL, fiber, and LTE broadband access with integrated voice services, security, routing, and business applications like IP PBX.
Acme Packet Presentation Materials for VUC June 18th 2010Michael Graves
1) The document discusses Acme Packet's enterprise session border controller (SBC) solutions which control four IP network borders, including SIP trunking, private networks, public internet, and hosted services.
2) It provides an overview of Acme Packet's SBC product portfolio including the Net-Net product family and their session capacity, throughput, and features for securing SIP trunking and enabling interoperability.
3) The SBC helps secure SIP trunking by acting as an application layer gateway, providing dynamic port control, full SIP firewalling, and DDOS protection to establish a "defense in depth" security model for SIP trunk traffic.
This document summarizes the key components needed to successfully implement IP video conferencing on a network, including quality of service, firewalls, gatekeepers, codecs, directory services, and registration/scheduling. It provides recommendations on setting up these elements and addresses common issues that can cause video calls to fail if not configured properly.
The document discusses issues and recommendations for implementing IP video conferencing on a state network. It provides background on the transition from ATM video to H.323 IP video. It recommends specific video codec hardware and software, addresses quality of service challenges, and recommends Click To Meet directory services and a dialing plan for the network. An IP Video Task Force was formed to address these issues and help with the implementation.
The document discusses CoreStor, an IP recording solution from Delma that can capture and record IP traffic, including VoIP packets. It describes various methods for capturing IP traffic, such as using span ports, port mirroring, conferencing, or custom gateways. CoreStor is designed to integrate seamlessly into existing systems and provide recording in a single chassis. It supports standard computer hardware and includes replay, administration, and analysis client software.
The document provides an overview of VoIP components, standards, architectures and implementation choices. It discusses key VoIP elements like terminals, packetized voice, soft switches, media servers, gateways, LANs/WANs and standards. It also describes common VoIP architectures for computer-computer, computer-phone, phone-phone communication over the internet. Finally, it outlines VoIP solutions for businesses using VoIP-enabled PBXs, replacing PBXs with softswitches, and hosted PBX solutions.
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.
The document discusses the features of Aethra Telecommunications' integrated access devices and IP PBX system. The devices provide broadband access, voice and fax ports, and advanced services. The IP PBX system offers standard features like call forwarding and transfer, as well as advanced features with the Micro Unified Communications license, including interactive voice response, voice mail, and call logs for billing. The system supports analog, ISDN, DECT and SIP phones and provides a comprehensive set of features to enable IP PBX and unified communications services for small businesses.
This document provides an overview of integrated access devices from Aethra Telecommunications, including broadband access options, voice and data port configurations, operating system features, and advanced application capabilities. Key products highlighted are the BG and SV series, which support ADSL, VDSL, SHDSL, fiber, and LTE broadband access with integrated voice services, security, routing, and business applications like IP PBX.
Acme Packet Presentation Materials for VUC June 18th 2010Michael Graves
1) The document discusses Acme Packet's enterprise session border controller (SBC) solutions which control four IP network borders, including SIP trunking, private networks, public internet, and hosted services.
2) It provides an overview of Acme Packet's SBC product portfolio including the Net-Net product family and their session capacity, throughput, and features for securing SIP trunking and enabling interoperability.
3) The SBC helps secure SIP trunking by acting as an application layer gateway, providing dynamic port control, full SIP firewalling, and DDOS protection to establish a "defense in depth" security model for SIP trunk traffic.
This document summarizes the key components needed to successfully implement IP video conferencing on a network, including quality of service, firewalls, gatekeepers, codecs, directory services, and registration/scheduling. It provides recommendations on setting up these elements and addresses common issues that can cause video calls to fail if not configured properly.
The document discusses issues and recommendations for implementing IP video conferencing on a state network. It provides background on the transition from ATM video to H.323 IP video. It recommends specific video codec hardware and software, addresses quality of service challenges, and recommends Click To Meet directory services and a dialing plan for the network. An IP Video Task Force was formed to address these issues and help with the implementation.
Reduce phone costs, improve connectivity, and secure your telecom and network equipment with Multi-Link’s ACP Series 2.0, Out-of-Band Network Switch & Call Router.
EntVoice is a software appliance that integrates various tools into a single interface to provide a complete voice over IP (VoIP) telephony system based on Asterisk. It offers features such as video calling, voicemail, conferencing, fax server capabilities, and reporting. The goal is to provide a reliable, modular, and easy to use open source telephony solution. Hardware appliance configurations are also available to run the EntVoice software.
The document provides a summary of Level 3's extensive portfolio of infrastructure, transport, IP, content delivery, Ethernet, voice, and enhanced network services. It describes the various features and benefits of their products and services, which include colocation, private line, wavelength, internet access, CDN, video transmission, VPN, router management, and voice termination offerings. Level 3 aims to deliver global connectivity with local flexibility through ongoing investment in their network and service capabilities.
The document summarizes the Allworx phone system, including its key features and capabilities. It was founded in 1998 and acquired by PAETEC in 2007. It provides IP phone systems for small and medium sized businesses with less than 100 users per location. The systems offer features like voicemail, conferencing, call queues, mobility applications, and support for multiple sites. They are designed to provide a full-featured phone system with a low total cost of ownership.
The document describes the Dana Server, a digital audio network appliance with a flexible I/O configuration. It has a 32-bit floating point DSP, analog and digital I/O cards, audio processing plugins, and network control capabilities. The Dana Server is designed to leverage existing IT infrastructure by using Ethernet for both audio and control data transport, providing greater scalability and simplicity for audio distribution applications.
The document describes the SSI5200, an indoor WiMAX subscriber station from SR Telecom. It provides:
- Fixed or nomadic access to voice, internet, and multimedia services through a compact and integrated design suitable for residential or small business use.
- Quick and easy installation via a web-based interface and integrated signal quality display for end users or professionals.
- Support for the most advanced WiMAX standards to provide a full range of services in a cost-effective unit.
iPECS-LIK is an IP communication solution from Ericsson-LG designed for small and medium sized businesses with 20 to 1,000 users. It brings together voice, messaging, video and mobile applications to increase productivity while delivering full PBX functionality. The distributed architecture allows connection of remote devices over IP networks and system networking of up to 250 systems. It employs a modular design to grow with business needs and a centralized management system for easy administration.
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 document provides an overview of key concepts related to Voice over IP (VoIP) technology. It defines common VoIP terms and standards, describes how VoIP works by breaking analog voice signals into digital packets, and outlines typical system elements like softswitches, terminals, and gateways. It also discusses media standards, signaling protocols, quality of service measures, fax transmission methods, and various Patton Electronics VoIP products.
The GENBAND 8840 IP Telephone is designed to boost productivity with high-end functionality for knowledge workers and administrative staff. It features wideband voice quality for comfortable calling and a customizable interface with programmable buttons. Additional benefits include flexible deployment through WiFi connectivity, optional Bluetooth compatibility, and remote provisioning to reduce management costs. The phone provides both advanced features and a design suited for contemporary offices.
This case study showcases Mistral’s capability in designing a flexible
and easy to upgrade VoIP Radio Gateway solution, interoperable with
all types of radio communications including conventional radios,
TETRA and TETRAPOL terminals. This IP Radio Gateway solution is
an exclusive and protected design for Mistral’s European customer,
Amper.
BT Etherflow is a managed wide area Ethernet service that provides flexible "building blocks" to meet changing network needs. It offers high bandwidth connectivity through BT's 21st Century Network, along with features like native Ethernet, access speeds up to 1Gbps, virtual connections between sites, and a customer portal for online management. BT Etherflow is designed for data-intensive environments that require high bandwidth, control over IP architecture, and support for non-IP applications.
XO IP Flex is a converged voice over IP solution that offers voice and data services through a single package for a flat monthly rate. It includes local and long distance calling, internet access, web hosting, and unified communications features. Customers can choose from port speeds and connectivity options. Standard features include unlimited local/site calling, a toll free number, web hosting, and call management tools. Optional applications like auto attendant, VPN, and call centers are also available.
The Polycom SoundPoint IP 450 is a mid-range SIP desktop phone that features Polycom HD Voice for high-quality audio, a high-resolution graphical display, and support for productivity applications through an XML microbrowser. It has a three-line LCD screen, 17 dedicated keys, and 4 soft keys. The phone provides clear transmission, integrated applications, and interoperability with SIP platforms.
The Skystar 360E is a satellite-based broadband IP solution for corporate networks that provides high-speed connectivity to dispersed business locations. It uses DVB standards and supports a wide range of IP applications. The system consists of a central hub and remote VSAT terminals connected via satellite. It offers benefits like centralized management, multicast capabilities, TCP acceleration, and interactive data and video conferencing to improve business efficiency and connectivity for SOHO, SME, and large corporate networks.
This document provides an overview and outline for designing a USB device driver. It discusses USB hardware controllers, the architecture of an embedded USB device including driver components and threads. It describes the USB device driver API including functions for initialization, opening/closing endpoints, reading/writing data, and handling control transfers. The document uses examples to illustrate interrupt handling, enumeration, and data transfer processes involving the USB controller hardware and endpoint FIFOs.
This document provides an overview of open source PBX software called Asterisk. It discusses VoIP technologies including codecs, protocols and PBX features. It also outlines how to install, configure and use Asterisk to set up a PBX system with channels, phones, IVRs and billing integration. Hardware requirements and options for interfaces are presented along with examples of configuration files. The document demonstrates how to register softphones and test calling between Asterisk and other VoIP systems.
The KX-NCV200 is a two-in-one call management system combining a Panasonic voice processing system with an ACD reporting system. The voice processing system provides features like voice mail, auto attendant, and email integration. The ACD reporting system provides real-time monitoring, performance reports, and call and agent information to help optimize call center management. The integrated system allows businesses to improve customer service and reduce telecommunications costs with versatile call routing and reporting tools.
The document provides configuration instructions for a Dialogic media gateway to integrate with the BlackBerry Mobile Voice System (MVS). It includes settings for IP, management protocols, routing tables, TDM, VoIP, DSP, and other postconfiguration tasks. Key settings include specifying the IP address of the BlackBerry MVS server, enabling HTTP/Telnet servers, configuring inbound and outbound routing rules to route calls between the TDM and VoIP networks, and configuring codecs, timers and other parameters.
The Panasonic KX-TDE Communications Platform is a robust and flexible IP communications system designed for businesses. It offers versatile features such as wireless capability, networking between multiple locations, and centralized voice mail. The system allows anytime, anywhere communication for mobile employees and helps businesses stay connected. It is scalable and can expand to support business growth. The system also includes productivity applications to enhance communications and collaboration.
Multipoint Video Conference Over Public InternetVideoguy
1) StarHub offers multipoint video conferencing over the public internet, which provides better quality than ISDN networks at a lower cost.
2) Using IP networks allows for higher quality video and audio through higher data rates compared to ISDN, and eliminates issues like lip synchronization problems.
3) The service offers reliability through a dedicated private IP network, ease of use with operator assistance, and affordability by avoiding call transport costs and supporting existing equipment.
Multipoint Video Conference Over Public InternetVideoguy
1) StarHub offers multipoint video conferencing over the public internet, which provides better quality than ISDN networks at a lower cost.
2) Video conferencing over the internet supports higher data rates on a single connection compared to ISDN, and uses existing internet infrastructure which is more cost effective than maintaining separate ISDN networks.
3) StarHub's solution provides operator assistance for conferencing, monitoring calls for disconnects, and reconnecting calls within moments if needed. This improves reliability, quality, and ease of use compared to ISDN conferencing.
Reduce phone costs, improve connectivity, and secure your telecom and network equipment with Multi-Link’s ACP Series 2.0, Out-of-Band Network Switch & Call Router.
EntVoice is a software appliance that integrates various tools into a single interface to provide a complete voice over IP (VoIP) telephony system based on Asterisk. It offers features such as video calling, voicemail, conferencing, fax server capabilities, and reporting. The goal is to provide a reliable, modular, and easy to use open source telephony solution. Hardware appliance configurations are also available to run the EntVoice software.
The document provides a summary of Level 3's extensive portfolio of infrastructure, transport, IP, content delivery, Ethernet, voice, and enhanced network services. It describes the various features and benefits of their products and services, which include colocation, private line, wavelength, internet access, CDN, video transmission, VPN, router management, and voice termination offerings. Level 3 aims to deliver global connectivity with local flexibility through ongoing investment in their network and service capabilities.
The document summarizes the Allworx phone system, including its key features and capabilities. It was founded in 1998 and acquired by PAETEC in 2007. It provides IP phone systems for small and medium sized businesses with less than 100 users per location. The systems offer features like voicemail, conferencing, call queues, mobility applications, and support for multiple sites. They are designed to provide a full-featured phone system with a low total cost of ownership.
The document describes the Dana Server, a digital audio network appliance with a flexible I/O configuration. It has a 32-bit floating point DSP, analog and digital I/O cards, audio processing plugins, and network control capabilities. The Dana Server is designed to leverage existing IT infrastructure by using Ethernet for both audio and control data transport, providing greater scalability and simplicity for audio distribution applications.
The document describes the SSI5200, an indoor WiMAX subscriber station from SR Telecom. It provides:
- Fixed or nomadic access to voice, internet, and multimedia services through a compact and integrated design suitable for residential or small business use.
- Quick and easy installation via a web-based interface and integrated signal quality display for end users or professionals.
- Support for the most advanced WiMAX standards to provide a full range of services in a cost-effective unit.
iPECS-LIK is an IP communication solution from Ericsson-LG designed for small and medium sized businesses with 20 to 1,000 users. It brings together voice, messaging, video and mobile applications to increase productivity while delivering full PBX functionality. The distributed architecture allows connection of remote devices over IP networks and system networking of up to 250 systems. It employs a modular design to grow with business needs and a centralized management system for easy administration.
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 document provides an overview of key concepts related to Voice over IP (VoIP) technology. It defines common VoIP terms and standards, describes how VoIP works by breaking analog voice signals into digital packets, and outlines typical system elements like softswitches, terminals, and gateways. It also discusses media standards, signaling protocols, quality of service measures, fax transmission methods, and various Patton Electronics VoIP products.
The GENBAND 8840 IP Telephone is designed to boost productivity with high-end functionality for knowledge workers and administrative staff. It features wideband voice quality for comfortable calling and a customizable interface with programmable buttons. Additional benefits include flexible deployment through WiFi connectivity, optional Bluetooth compatibility, and remote provisioning to reduce management costs. The phone provides both advanced features and a design suited for contemporary offices.
This case study showcases Mistral’s capability in designing a flexible
and easy to upgrade VoIP Radio Gateway solution, interoperable with
all types of radio communications including conventional radios,
TETRA and TETRAPOL terminals. This IP Radio Gateway solution is
an exclusive and protected design for Mistral’s European customer,
Amper.
BT Etherflow is a managed wide area Ethernet service that provides flexible "building blocks" to meet changing network needs. It offers high bandwidth connectivity through BT's 21st Century Network, along with features like native Ethernet, access speeds up to 1Gbps, virtual connections between sites, and a customer portal for online management. BT Etherflow is designed for data-intensive environments that require high bandwidth, control over IP architecture, and support for non-IP applications.
XO IP Flex is a converged voice over IP solution that offers voice and data services through a single package for a flat monthly rate. It includes local and long distance calling, internet access, web hosting, and unified communications features. Customers can choose from port speeds and connectivity options. Standard features include unlimited local/site calling, a toll free number, web hosting, and call management tools. Optional applications like auto attendant, VPN, and call centers are also available.
The Polycom SoundPoint IP 450 is a mid-range SIP desktop phone that features Polycom HD Voice for high-quality audio, a high-resolution graphical display, and support for productivity applications through an XML microbrowser. It has a three-line LCD screen, 17 dedicated keys, and 4 soft keys. The phone provides clear transmission, integrated applications, and interoperability with SIP platforms.
The Skystar 360E is a satellite-based broadband IP solution for corporate networks that provides high-speed connectivity to dispersed business locations. It uses DVB standards and supports a wide range of IP applications. The system consists of a central hub and remote VSAT terminals connected via satellite. It offers benefits like centralized management, multicast capabilities, TCP acceleration, and interactive data and video conferencing to improve business efficiency and connectivity for SOHO, SME, and large corporate networks.
This document provides an overview and outline for designing a USB device driver. It discusses USB hardware controllers, the architecture of an embedded USB device including driver components and threads. It describes the USB device driver API including functions for initialization, opening/closing endpoints, reading/writing data, and handling control transfers. The document uses examples to illustrate interrupt handling, enumeration, and data transfer processes involving the USB controller hardware and endpoint FIFOs.
This document provides an overview of open source PBX software called Asterisk. It discusses VoIP technologies including codecs, protocols and PBX features. It also outlines how to install, configure and use Asterisk to set up a PBX system with channels, phones, IVRs and billing integration. Hardware requirements and options for interfaces are presented along with examples of configuration files. The document demonstrates how to register softphones and test calling between Asterisk and other VoIP systems.
The KX-NCV200 is a two-in-one call management system combining a Panasonic voice processing system with an ACD reporting system. The voice processing system provides features like voice mail, auto attendant, and email integration. The ACD reporting system provides real-time monitoring, performance reports, and call and agent information to help optimize call center management. The integrated system allows businesses to improve customer service and reduce telecommunications costs with versatile call routing and reporting tools.
The document provides configuration instructions for a Dialogic media gateway to integrate with the BlackBerry Mobile Voice System (MVS). It includes settings for IP, management protocols, routing tables, TDM, VoIP, DSP, and other postconfiguration tasks. Key settings include specifying the IP address of the BlackBerry MVS server, enabling HTTP/Telnet servers, configuring inbound and outbound routing rules to route calls between the TDM and VoIP networks, and configuring codecs, timers and other parameters.
The Panasonic KX-TDE Communications Platform is a robust and flexible IP communications system designed for businesses. It offers versatile features such as wireless capability, networking between multiple locations, and centralized voice mail. The system allows anytime, anywhere communication for mobile employees and helps businesses stay connected. It is scalable and can expand to support business growth. The system also includes productivity applications to enhance communications and collaboration.
Multipoint Video Conference Over Public InternetVideoguy
1) StarHub offers multipoint video conferencing over the public internet, which provides better quality than ISDN networks at a lower cost.
2) Using IP networks allows for higher quality video and audio through higher data rates compared to ISDN, and eliminates issues like lip synchronization problems.
3) The service offers reliability through a dedicated private IP network, ease of use with operator assistance, and affordability by avoiding call transport costs and supporting existing equipment.
Multipoint Video Conference Over Public InternetVideoguy
1) StarHub offers multipoint video conferencing over the public internet, which provides better quality than ISDN networks at a lower cost.
2) Video conferencing over the internet supports higher data rates on a single connection compared to ISDN, and uses existing internet infrastructure which is more cost effective than maintaining separate ISDN networks.
3) StarHub's solution provides operator assistance for conferencing, monitoring calls for disconnects, and reconnecting calls within moments if needed. This improves reliability, quality, and ease of use compared to ISDN conferencing.
Multipoint Video Conference Over Public InternetVideoguy
1) StarHub offers multipoint video conferencing over the public internet, which provides better quality than ISDN networks at a lower cost.
2) Their solution uses a single internet connection rather than multiple ISDN lines, eliminating issues like line failures and ensuring a consistent connection quality.
3) Hosting video conferences over the internet rather than ISDN networks saves on usage fees, making it a more affordable option, while still supporting high call rates without additional bandwidth costs.
VoIP allows users to make phone calls using an Internet connection rather than a traditional phone line. It works by converting the voice signal from analog to digital, breaking it into packets, sending it over IP, reassembling it at the destination, and converting it back to analog. VoIP has advantages like low cost and portability but disadvantages like quality issues during power outages or network instability. Major challenges include addressing latency, echo, jitter, connection problems through firewalls and NAT, and overall reliability.
This article introduces a view of a generic Service Provider IP distribution system including DVB's IP standard; a comparison of Internet and managed SP IP video distribution; how a broadcaster can inject TV programming into the Internet and, finally, how to control the Quality of Experience of video in an IP network.
DHCP automates the assignment of IP addresses, subnet masks, gateways and other network parameters to clients on a network from a central server, making management of a large network easier. NAT allows an internal network using private IP addresses to access the public internet by translating those private addresses to a public IP address, with translation done either statically or dynamically by a device like a broadband router. Port address translation is a variation of NAT that allows external hosts to access specific internal hosts by forwarding certain traffic to the internal host's private IP and port.
This document discusses how VoIP systems work, including:
1) Soft switches translate phone numbers to IP addresses and know the current location and IP address of endpoints on the network.
2) Common VoIP protocols like H.323 and SIP are used to connect hardware and support real-time video, audio, and data applications. However, a lack of standardization can cause compatibility issues.
3) VoIP offers advantages over traditional phone systems like lower costs, mobility via internet connections, and reduced bandwidth requirements.
This document discusses prospects for managed and hosted voice services. It identifies over 200 work products involved in voice services that could be managed, such as call detail recording, trunk configuration analysis, and fraud detection. It describes how to select a support model based on whether the service needs to be on-site, remote, or mobile. Finally, it outlines some key components of managed voice services, including fault management, configuration management, and security management.
This document provides an overview of Voice over Internet Protocol (VoIP) technology. It describes how VoIP works by converting voice signals to digital data that is transmitted over the Internet using packet switching. Common VoIP protocols like SIP and H.323 are discussed along with VoIP components like softphones, gateways, and codecs. Advantages of VoIP include low cost and flexibility, while disadvantages include reliability issues and lack of service during power outages. The document recommends that most VoIP issues will be addressed by 2008 when it will gain widespread consumer acceptance.
Webinar de la marca Patton explicando las funcionalidades de los SBC ESBR y escenarios de uso
All IP Networks – EOL for ISDN
Survivability, Service Demarcation, QoS
VoIP Security
VoIP Encryption, SIP TLS SRTP, Network topology hiding
IP Routing Performance
IP Routing Performance for multi service application
Ease of use
Simple configuration, deployment
Pricing
Price-value ratio
Cisco CallManager Express (CME) is a call processing solution that provides VoIP functionality for small to medium sized networks of up to 120 IP phones. It allows connection to the PSTN via analog or digital trunks and supports protocols like Skinny and H.323 for call control. CME is configured on Cisco IOS routers and gateways to provide integrated voice and data services over IP.
Why Session Border Controllers?
Product Portfolio of the Session Border Controller
Business Applications and Use Cases (Vega ESBC)
Carrier/Service Provider Applications and Use Cases (NetBorder SBC)
Sangoma SBC Load Balancing and Failover Techniques
SBC Walkthrough
Conceptual Overview of the SBC Call Processing Components
Introduction and Configuration of SIP Profiles
Introduction and Configuration of Domain Profiles
Introduction and Configuration of Media Profiles
Introduction and Configuration of SIP Trunks
Introduction and Configuration of Call Routing
Walkthrough
Video conferencing allows people at different physical locations to conduct face-to-face meetings virtually. It works by using computer networks and audio-visual equipment to transmit video and audio data between two or more locations in real-time. Key components of video conferencing systems include video cameras, microphones, screens or monitors, speakers, a codec to compress and decompress the audio-visual data, and a network connection. Popular protocols for video conferencing include H.320 for ISDN networks and H.323 for internet-based video calls.
This document provides an overview of VoIP security. It discusses the basics of VoIP security including authentication, authorization, availability, and encryption. It outlines some common attack vectors such as accessing an unsecured local network connection, wireless network, or public network. It also mentions threats from compromising a phone's configuration file or uploading a malicious file. The document summarizes some unconventional VoIP security threats like phishing, caller ID spoofing, eavesdropping, call redirection, and spam over internet telephony.
you can be friend with me on orkut
"mangalforyou@gmail.com" : i belive in sharing the knowledge so please send project reports ,seminar and ppt. to me .
This document provides an overview of Internet Protocol Telephony (VoIP). It discusses how VoIP works by digitizing and compressing voice into packets transmitted over the Internet. It also covers some of the common protocols used, including Session Initiation Protocol (SIP) and H.323, and compares their advantages. Potential applications and challenges of VoIP are also mentioned.
This document provides an overview of remote networking deployments using Aruba Instant and remote access points. It defines remote APs as Aruba access points deployed at remote sites and plugged into a router connected to a modem. The document discusses the different modes remote APs can operate in, including tunnel mode where all traffic is forwarded through an IPsec tunnel, split-tunnel mode where corporate traffic uses the tunnel and local traffic is forwarded locally, and bridge mode where only control traffic uses the tunnel. It also provides an overview of the Aruba Instant architecture and features, such as the virtual controller, dynamic RADIUS proxy, guest access using a captive portal, mesh capabilities, and deployment guidelines.
This document discusses network address translation (NAT) features and how to design a secure NAT network. It describes different types of NAT including static, dynamic, and masquerading NAT. When implementing NAT, considerations include network size, security needs, location, IP addressing, and data flow rates. Securing the NAT network involves using routing/remote access filters, address pools, special ports, and VPN connections. The optimal design devotes one machine to act as the NAT server, connecting over persistent routes with multiple internet connections to enhance performance and availability.
The document discusses the open source Asterisk PBX software. It provides an overview of Asterisk including that it was created in 1999 as a free and open source alternative to expensive proprietary PBX systems. Asterisk allows users to build their own software-based phone systems using inexpensive hardware and can provide many of the same features as traditional PBXs through its flexible architecture and extensive capabilities. The document outlines some of Asterisk's main functionalities and how it works as well as hardware that can be used with it.
This paper proposes an adaptive energy management policy for wireless video streaming between a battery-powered client and server. It models the energy consumption of the server and client based on factors like CPU frequency, transmission power, and channel bandwidth. The paper formulates an optimization problem to assign optimal energy to each video frame. This maximizes system lifetime while meeting a minimum video quality requirement. Experimental results show the proposed policy increases overall system lifetime by 20% on average.
Microsoft PowerPoint - WirelessCluster_PresVideoguy
This document analyzes delays in unicast video streaming over IEEE 802.11 WLAN networks. It describes conducting an experiment using a testbed with a Darwin Streaming Server and WLAN probe to capture packets. The analysis found that video bitrate variations, packetization scheme, bandwidth load, and frame-based nature of video all impacted mean delay. Bursts of packets from video frames caused per-packet delay to increase in a sawtooth pattern. Increasing uplink load was also found to affect delay variations.
Proxy Cache Management for Fine-Grained Scalable Video StreamingVideoguy
This document proposes a novel video caching framework that uses MPEG-4 Fine-Grained Scalable (FGS) video with post-encoding rate control to achieve low-cost and fine-grained rate adaptation. The framework allows clients to have heterogeneous bandwidths and enables adaptive control of backbone bandwidth consumption. It examines issues in caching FGS videos, such as determining the optimal portion to cache (in terms of length and rate) and optimal streaming rate to clients. Simulation results show it significantly reduces transmission costs compared to non-adaptive caching while providing flexible utility to heterogeneous clients with low computational overhead.
The document compares Microsoft Windows Media and the Adobe Flash Platform for streaming media. It discusses key differences like user experience, workflows, and playback reach. Flash offers more flexibility in creative expression, richer interactions, and wider device playback than Windows Media. It also has a 98% install base, making it easier for viewers to watch streams without extra software. The document outlines workflows for experience design, programming, broadcasting, production, and more using Flash tools versus Microsoft alternatives.
Free-riding Resilient Video Streaming in Peer-to-Peer NetworksVideoguy
This document summarizes a PhD thesis about free-riding resilient video streaming in peer-to-peer networks. The thesis contains research on two approaches: tree-based live streaming and swarm-based video-on-demand. For tree-based live streaming, the thesis presents the Orchard algorithm for constructing and maintaining trees to distribute video in a peer-to-peer network. It analyzes attacks on Orchard like free-riding and evaluates Orchard's performance under different conditions through experiments. For swarm-based video-on-demand, the thesis introduces the Give-to-Get approach for distributing video files and compares it to other peer-to-peer protocols. It evaluates Give-to-Get's performance in experiments
BT has developed Fastnets technology to improve video streaming. It avoids start-up delays and picture freezing during congestion. Fastnets streams multiple encoded versions of the video at different data rates and seamlessly switches between them based on available bandwidth to maintain quality without pausing. This allows for near-instant start times and reduces bandwidth usage by up to 30%. Fastnets provides a high-quality video streaming solution for both mobile and IPTV applications.
This document summarizes recent research on video streaming over Bluetooth networks. It discusses three key areas: intermediate protocols, quality of service (QoS) control, and media compression. For intermediate protocols, it evaluates streaming via HCI, L2CAP, and IP layers and their tradeoffs. For QoS control, it describes how error control mechanisms like link layer FEC, retransmission, and error concealment can improve video quality over Bluetooth. It also discusses congestion control. For media compression, it notes the importance of compression to achieve efficiency over limited Bluetooth bandwidths.
The document discusses video streaming, including definitions and concepts. It covers topics such as the difference between streaming and downloading, common streaming categories like live and on-demand, protocols used for streaming like RTSP and RTP, and the development process for creating streaming video including content planning, capturing, editing, encoding, and integrating with servers.
Inlet Technologies offers a live video streaming solution called Spinnaker that uses Intel Xeon processors with quad-core technology. Spinnaker can encode live video streams into multiple formats and resolutions simultaneously. This allows content to be delivered optimally to various devices. Spinnaker is a flexible, scalable solution that can increase broadcast capacity cost-effectively while maintaining high video quality.
Considerations for Creating Streamed Video Content over 3G ...Videoguy
The document discusses considerations for creating video content that can be streamed over mobile networks with restricted bandwidth like 3G-324M. It covers topics like video basics, codecs, profiles and levels, video streaming techniques, guidelines for authoring mobile-friendly content, and tools for analyzing video streams. The goal is to help content creators optimize video quality for low-bandwidth mobile viewing.
ADVANCES IN CHANNEL-ADAPTIVE VIDEO STREAMINGVideoguy
This document summarizes recent advances in channel-adaptive video streaming. It reviews adaptive media playout at the client to reduce latency, rate-distortion optimized packet scheduling to determine the best packet to send, and channel-adaptive packet dependency control to improve error robustness and reduce latency. It also discusses challenges for wireless video streaming and different wireless streaming architectures.
Impact of FEC Overhead on Scalable Video StreamingVideoguy
The document discusses the impact of forward error correction (FEC) overhead on scalable video streaming. It aims to address uncertainty about the benefits of FEC and provide insight into how FEC overhead affects scalable video performance. The motivation section explains that FEC is often used for streaming to overcome packet loss without retransmission. However, previous studies have reported conflicting results on the benefits of FEC. The background section provides details on media-independent FEC schemes.
The document proposes a cost-effective solution for video streaming and rich media applications using Vela's RapidAccess video server combined with iQstor's iQ1200 SATA storage system. The integrated encoding, decoding and video serving capabilities of RapidAccess are paired with the scalable storage and virtualization features of the iQ1200 SATA storage array to provide a robust yet affordable infrastructure for applications such as video on demand, corporate training and distance learning.
This document provides information on streaming video into Second Life, including:
- The basic prerequisites for streaming video include being the landowner, using QuickTime format videos, and having the video hosted on a web server.
- There are three main ways to stream video: establishing movie playback, streaming live video, and broadcasting from Second Life.
- Streaming live video or broadcasting involves using software like QuickTime Broadcaster or Windows Media Encoder to capture the video stream and send it to a hosting server, then entering that URL in Second Life.
XStream Live 2 is a live video encoding and streaming software that allows users to broadcast high quality HD video at low bitrates. It supports various video formats and streaming servers. The software provides high quality H.264 encoding with proprietary technology. It is designed for live event streaming, IPTV, and other video distribution uses.
The document provides instructions for setting up a homemade videoconference streaming solution using Windows Media software. The solution involves installing Windows Media Encoder and Administrator on a server and configuring the software to receive a video stream from a videoconferencing terminal. The streaming server then broadcasts the stream in real-time to clients who can view it using media player software. The solution provides a low-cost way to stream videoconferences but has limitations such as only supporting one conference stream at a time.
This document describes iStream Live 2 software for live streaming video to iPhones and iPads. It allows streaming of SD or HD video over HTTP from a variety of video sources. Key features include support for all major CDNs, encoding of H.264 video and AAC audio for high quality at low bitrates, and integration with existing Windows streaming systems. It provides better quality streaming than other encoders at lower bandwidth requirements.
Glow: Video streaming training guide - FirefoxVideoguy
This document provides a guide to using Glow video streaming. It includes tutorials on setting up video streaming by adding the Video Streaming Management web part, uploading video clips, viewing clips, editing clip information, and deleting clips. The guide also discusses how video streaming can be used to support learning and teaching, such as adding videos to lessons.
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