- The document discusses implementing cost-efficient H.320 video conferencing over ISDN networks, including multipoint conferencing. It describes the components needed for H.320 conferencing and alternatives to reduce costs, such as using a multipoint control unit bureau service or least cost call routing. The document provides an overview of ISDN, dedicated multipoint control units, endpoints with embedded multipoint capabilities, and using an MCU bureau service to reduce capital expenditures.
What you really need to know about Video Conferencing SystemsVideoguy
This document discusses factors to consider when choosing a video conferencing system, including available bandwidth, acceptable quality levels, and supported standards. It outlines different connection types like ISDN, LAN/WAN, cellular networks and their associated standards. Newer standards like H.264 can provide better quality at lower bandwidths. The best system depends on expectations, bandwidth, number of participants, locations, management needs and costs.
This document summarizes standards and technologies for videoconferencing, including H.323, H.320, H.264, and others. It describes protocols like H.245 and RTP, as well as elements of an H.323 system like terminals, multipoint control units (MCUs), and gatekeepers. It provides information on audio and video codecs, transport media, directory services, and cellular/3G extensions.
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.
1) Videoconferencing allows participants to see, hear and collaborate in real-time over networks or the internet. It requires equipment like cameras, microphones and displays.
2) Key standards like H.320, H.323 and SIP define how audio, video and data are transmitted over different networks like ISDN, IP and cellular. Codecs compress video and audio for efficient transmission.
3) Popular applications of videoconferencing include meetings, education, telemedicine and more. Proper etiquette like preparing agendas, camera positioning and avoiding distractions enhances collaboration.
1) Videoconferencing allows participants to see, hear and collaborate in real time over networks or the internet. It requires equipment like cameras, microphones and displays.
2) Standards like H.320, H.323 and H.324 define protocols for videoconferencing over different mediums. Codecs compress audio and video for transmission. Transport protocols include TCP, UDP and RTP.
3) Popular applications of videoconferencing include meetings, education, telemedicine and more. Setup, quality and costs vary depending on the medium used such as ISDN, IP networks or cellular networks.
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 study guide is intended to provide those pursuing the CCNA certification with a framework of what concepts need to be studied. This is not a comprehensive document containing all the secrets of the CCNA, nor is it a “braindump” of questions and answers.
I sincerely hope that this document provides some assistance and clarity in your studies.
Integrated Services Digital Network (ISDN) is a set of communication standards that allows digital transmission and management of different types of data such as voice, text, and video over either digital or analog network infrastructures. ISDN provides end-to-end digital connectivity to support plain old telephone service (POTS) as well as a variety of data services. It defines several interface standards and protocols to establish and maintain digital communication channels.
What you really need to know about Video Conferencing SystemsVideoguy
This document discusses factors to consider when choosing a video conferencing system, including available bandwidth, acceptable quality levels, and supported standards. It outlines different connection types like ISDN, LAN/WAN, cellular networks and their associated standards. Newer standards like H.264 can provide better quality at lower bandwidths. The best system depends on expectations, bandwidth, number of participants, locations, management needs and costs.
This document summarizes standards and technologies for videoconferencing, including H.323, H.320, H.264, and others. It describes protocols like H.245 and RTP, as well as elements of an H.323 system like terminals, multipoint control units (MCUs), and gatekeepers. It provides information on audio and video codecs, transport media, directory services, and cellular/3G extensions.
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.
1) Videoconferencing allows participants to see, hear and collaborate in real-time over networks or the internet. It requires equipment like cameras, microphones and displays.
2) Key standards like H.320, H.323 and SIP define how audio, video and data are transmitted over different networks like ISDN, IP and cellular. Codecs compress video and audio for efficient transmission.
3) Popular applications of videoconferencing include meetings, education, telemedicine and more. Proper etiquette like preparing agendas, camera positioning and avoiding distractions enhances collaboration.
1) Videoconferencing allows participants to see, hear and collaborate in real time over networks or the internet. It requires equipment like cameras, microphones and displays.
2) Standards like H.320, H.323 and H.324 define protocols for videoconferencing over different mediums. Codecs compress audio and video for transmission. Transport protocols include TCP, UDP and RTP.
3) Popular applications of videoconferencing include meetings, education, telemedicine and more. Setup, quality and costs vary depending on the medium used such as ISDN, IP networks or cellular networks.
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 study guide is intended to provide those pursuing the CCNA certification with a framework of what concepts need to be studied. This is not a comprehensive document containing all the secrets of the CCNA, nor is it a “braindump” of questions and answers.
I sincerely hope that this document provides some assistance and clarity in your studies.
Integrated Services Digital Network (ISDN) is a set of communication standards that allows digital transmission and management of different types of data such as voice, text, and video over either digital or analog network infrastructures. ISDN provides end-to-end digital connectivity to support plain old telephone service (POTS) as well as a variety of data services. It defines several interface standards and protocols to establish and maintain digital communication channels.
This document provides an overview of broadband and VoIP (Voice over Internet Protocol). It discusses key VoIP components like terminals, gateways, gatekeepers, and MCUs. It also examines the H.323 and SIP protocols used in VoIP. Finally, it outlines benefits of VoIP and requirements like broadband internet connectivity. The document aims to inform readers about fundamental concepts and technologies relating to broadband and VoIP.
H.323 Network Components include H.323 Terminals, Gatekeepers ...Videoguy
The document summarizes the key components of an H.323 network: H.323 terminals, gatekeepers, gateways, and multipoint control units (MCUs). It describes the functions of each component, including how H.323 terminals communicate via standards-defined protocols, how gatekeepers provide address translation and call control, how gateways allow interoperability between H.323 and H.320 systems, and how MCUs allow multiparty video conferences. It also provides examples of how these components work together to establish calls within an H.323 network.
The document contains information about several individuals and an outline for a presentation on H.323. The outline discusses what H.323 is, its scope and importance, its historical development stages, the elements that make up an H.323 system, the core protocols that define H.323 communication, how H.323 calls are signaled, and the future prospects of H.323.
Implementing QoS Solutions for H.323 Video Conferencing over IPVideoguy
This document discusses implementing Quality of Service (QoS) solutions for H.323 video conferencing over IP networks. It begins by outlining the prerequisites for understanding H.323 protocols and components. It then provides background on H.323 and characterizes video conference traffic. The document describes how to plan network capacity and determine per-call bandwidth needs. It recommends classifying and prioritizing traffic using DiffServ codes and queues like Low Latency Queueing (LLQ). Sample configurations are provided to shape traffic and interwork H.323 terminals with QoS.
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.
The document discusses videoconferencing standards defined by the ITU, including H.323, H.320, audio standards like G.711, video standards like H.261 and H.263, communication standards, and T.120 for data collaboration. It provides an overview of key standards, their components and functions, such as terminals, gatekeepers, and gateways in H.323.
Network Planning Worksheets for Video ConferencingVideoguy
This document contains worksheets to help order ISDN for video conferencing systems. It includes sections to gather information needed to order ISDN in the US and Canada or outside of these regions. It also has a section on preparing a video conferencing system for installation on a LAN computer network. The worksheets should be completed with an administrator and/or ISDN provider to ensure the correct ISDN line is provisioned for the video conferencing system. Additional resources on preparing networks for video conferencing are referenced.
Designing Triple-Play Apps Using DSP Resource BoardsVideoguy
The document discusses the optimal hardware and software architectures for designing triple-play applications using DSP resource boards. It recommends using powerful DSPs with external memory that can handle all media types on a single hardware platform. The software should have an open framework with flexible APIs and remote diagnostics to support new features and algorithms from multiple vendors. The media and control paths should be separate to avoid bottlenecks and reduce host processor load.
VoIP uses packet networks to carry voice calls in addition to data. It works by converting analog voice signals to digital data packets which are transmitted over IP networks and reconverted to analog at the receiving end. Key components include IP phones, signaling servers, and protocols like SIP and H.323 which handle call setup and signaling. Quality of service for VoIP depends on factors like packet loss, delay, and jitter which can be managed through queuing and reserving bandwidth for voice traffic.
This document provides an overview of InfiniBand, including:
- InfiniBand is an architecture and specification that promises greater bandwidth and expandability compared to PCI, with throughput up to 2.5 gigabytes per second and support for up to 64,000 devices.
- It uses a switched fabric architecture rather than a shared bus, allowing high bandwidth, low latency connections between nodes over long distances using fiber optic or copper cables.
- The key components are host channel adapters, target channel adapters, switches, and routers. Data is transmitted between nodes via packets routed through the fabric.
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.
Demystifying Multimedia Conferencing Over the Internet Using ...Videoguy
The document provides an overview of the H.323 standards for multimedia conferencing over packet networks:
- H.323 defines terminals, equipment, and services for real-time audio, video, and data conferencing over networks like the Internet.
- Key components include terminals, gatekeepers, gateways, and multipoint control units. Terminals establish calls using Q.931 signaling and exchange capabilities with H.245.
- The document describes the basic call flow between two terminals without a gatekeeper, including capability exchange and logical channel establishment. It also outlines call signaling with a gatekeeper involved.
VoIP is the acronym for Voice over Internet Protocol, which means telephone services over the Internet. It works by converting voice data into packets that can be delivered over IP networks. Some key protocols used for VoIP include H.323, SIP, SDP, MGCP, and RTP. H.323 defines components and procedures for real-time multimedia sessions over packet networks, including audio, video, and data. SIP establishes, modifies, and terminates multimedia sessions like IP telephony by registering systems and providing SIP addresses to check availability and invite others to sessions.
Video Teleconferencing (VTC) Technology at the National ...Videoguy
This document provides information about video teleconferencing (VTC) technology standards and practices at the National Institutes of Health (NIH). It discusses the business objectives for a standardized VTC architecture, including enabling firewall passage, remote management, standard naming conventions, dedicated telephony services, and compliance with ITU standards. It also describes the current VTC infrastructure and practices at NIH, including support for both internet protocol (IP) and integrated services digital network (ISDN) videoconferencing according to ITU recommendations H.323 and H.320. Finally, it lists the compliant specifications for VTC endpoints to integrate with the NIH infrastructure.
- Videoconferencing allows participants to see, hear and collaborate in real time over internet or telephone networks. It requires equipment like cameras, microphones, displays and codecs to compress and decompress audio/video data.
- Standards like H.320, H.323 and H.324 specify protocols for videoconferencing over ISDN, IP networks and POTS lines. Transport methods include ISDN, IP networks, cellular networks and POTS lines.
- Key components of videoconferencing systems are video/audio input/output devices, data transfer networks, and codecs. Formats like H.261, H.263, H.264 and audio standards G.711, G.722 are
This document summarizes key concepts and standards related to videoconferencing. It discusses audio teleconferencing, videoconferencing, and web conferencing. It defines videoconferencing as communication across long distances with video and audio using digital video transmission systems. It describes H.320 and H.323 as the main videoconferencing standards, with H.320 using ISDN lines and H.323 using packet-based networks like LANs. It also defines common videoconferencing terms like ISDN, codecs, bridges, full-duplex audio, and full-motion video.
The document provides an overview of videoconferencing basics, including what videoconferencing is, its benefits, how it is being used, common system components, and top manufacturers. It also discusses network connections, protocols, and different types of conferencing such as point-to-point, multipoint, webcasting, web conferencing, and audio conferencing.
The document summarizes INTEGRIS's video conferencing infrastructure and plans. It describes their transition from dedicated H.320 connections to an IP-based system using H.323 and H.264 standards across their WAN. Their current Polycom bridge supports 24 sites at 384kbps or 12 sites at 768kbps. Upgrading the bridge would allow mixing newer H.264 endpoints with older H.263 ones. Newer desktop and all-in-one systems offer improved video quality but require additional bandwidth or an upgrade to fully utilize the H.264 standard.
H.323 is the standard for multimedia conferences over IP networks. This document discusses implementing QoS solutions for H.323 video conferencing over an enterprise WAN with low-speed links. It provides guidance on capacity planning, determining bandwidth needs, classifying and prioritizing traffic. The recommended approach is to use DSCP values to classify voice as EF, video as AF41, and control traffic as AF31. Queuing mechanisms like LLQ and CBWFQ are suggested to provide minimum bandwidth guarantees for real-time traffic.
This document provides information about HEAnet's video conferencing service, which allows institutions to conduct video conferences over IP networks using H.323 standards. It describes the basic principles and elements of H.323 video conferencing. To use the service, an institution needs an H.323 compatible terminal and a connection to HEAnet's network. The document outlines how to register a terminal with HEAnet's gatekeeper to obtain a Global Dialing Scheme (GDS) number and make video calls within the HEAnet network or to external connections. Security considerations for opening necessary ports on an institution's firewall are also discussed.
This document provides an overview of multimedia services over IP networks and discusses two key protocols used: SIP and H.323. It describes the basics of SIP including session descriptions using SDP, message format, and session initiation. It also discusses SIP applications like IMS including requirements, protocols used, and architecture. For H.323, it outlines the network architecture including terminals, MCUs, gateways, and gatekeepers. It then describes the H.323 signaling protocols including RAS, H.225 call signaling, and H.245 call control.
This document provides an outline on voice over internet protocol (VoIP) covering topics such as how VoIP works, advantages of VoIP, types of codecs used for converting analog signals to digital data, signaling protocols like H.323 and SIP used for setting up and managing calls, and security considerations in VoIP like the use of SRTP for encryption. The document compares VoIP to traditional PSTN telephone networks and PBX systems, and explains that VoIP uses packet switching over the internet to make phone calls instead of dedicated circuits, allowing for upgrades with only increased bandwidth.
This document provides an overview of broadband and VoIP (Voice over Internet Protocol). It discusses key VoIP components like terminals, gateways, gatekeepers, and MCUs. It also examines the H.323 and SIP protocols used in VoIP. Finally, it outlines benefits of VoIP and requirements like broadband internet connectivity. The document aims to inform readers about fundamental concepts and technologies relating to broadband and VoIP.
H.323 Network Components include H.323 Terminals, Gatekeepers ...Videoguy
The document summarizes the key components of an H.323 network: H.323 terminals, gatekeepers, gateways, and multipoint control units (MCUs). It describes the functions of each component, including how H.323 terminals communicate via standards-defined protocols, how gatekeepers provide address translation and call control, how gateways allow interoperability between H.323 and H.320 systems, and how MCUs allow multiparty video conferences. It also provides examples of how these components work together to establish calls within an H.323 network.
The document contains information about several individuals and an outline for a presentation on H.323. The outline discusses what H.323 is, its scope and importance, its historical development stages, the elements that make up an H.323 system, the core protocols that define H.323 communication, how H.323 calls are signaled, and the future prospects of H.323.
Implementing QoS Solutions for H.323 Video Conferencing over IPVideoguy
This document discusses implementing Quality of Service (QoS) solutions for H.323 video conferencing over IP networks. It begins by outlining the prerequisites for understanding H.323 protocols and components. It then provides background on H.323 and characterizes video conference traffic. The document describes how to plan network capacity and determine per-call bandwidth needs. It recommends classifying and prioritizing traffic using DiffServ codes and queues like Low Latency Queueing (LLQ). Sample configurations are provided to shape traffic and interwork H.323 terminals with QoS.
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.
The document discusses videoconferencing standards defined by the ITU, including H.323, H.320, audio standards like G.711, video standards like H.261 and H.263, communication standards, and T.120 for data collaboration. It provides an overview of key standards, their components and functions, such as terminals, gatekeepers, and gateways in H.323.
Network Planning Worksheets for Video ConferencingVideoguy
This document contains worksheets to help order ISDN for video conferencing systems. It includes sections to gather information needed to order ISDN in the US and Canada or outside of these regions. It also has a section on preparing a video conferencing system for installation on a LAN computer network. The worksheets should be completed with an administrator and/or ISDN provider to ensure the correct ISDN line is provisioned for the video conferencing system. Additional resources on preparing networks for video conferencing are referenced.
Designing Triple-Play Apps Using DSP Resource BoardsVideoguy
The document discusses the optimal hardware and software architectures for designing triple-play applications using DSP resource boards. It recommends using powerful DSPs with external memory that can handle all media types on a single hardware platform. The software should have an open framework with flexible APIs and remote diagnostics to support new features and algorithms from multiple vendors. The media and control paths should be separate to avoid bottlenecks and reduce host processor load.
VoIP uses packet networks to carry voice calls in addition to data. It works by converting analog voice signals to digital data packets which are transmitted over IP networks and reconverted to analog at the receiving end. Key components include IP phones, signaling servers, and protocols like SIP and H.323 which handle call setup and signaling. Quality of service for VoIP depends on factors like packet loss, delay, and jitter which can be managed through queuing and reserving bandwidth for voice traffic.
This document provides an overview of InfiniBand, including:
- InfiniBand is an architecture and specification that promises greater bandwidth and expandability compared to PCI, with throughput up to 2.5 gigabytes per second and support for up to 64,000 devices.
- It uses a switched fabric architecture rather than a shared bus, allowing high bandwidth, low latency connections between nodes over long distances using fiber optic or copper cables.
- The key components are host channel adapters, target channel adapters, switches, and routers. Data is transmitted between nodes via packets routed through the fabric.
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.
Demystifying Multimedia Conferencing Over the Internet Using ...Videoguy
The document provides an overview of the H.323 standards for multimedia conferencing over packet networks:
- H.323 defines terminals, equipment, and services for real-time audio, video, and data conferencing over networks like the Internet.
- Key components include terminals, gatekeepers, gateways, and multipoint control units. Terminals establish calls using Q.931 signaling and exchange capabilities with H.245.
- The document describes the basic call flow between two terminals without a gatekeeper, including capability exchange and logical channel establishment. It also outlines call signaling with a gatekeeper involved.
VoIP is the acronym for Voice over Internet Protocol, which means telephone services over the Internet. It works by converting voice data into packets that can be delivered over IP networks. Some key protocols used for VoIP include H.323, SIP, SDP, MGCP, and RTP. H.323 defines components and procedures for real-time multimedia sessions over packet networks, including audio, video, and data. SIP establishes, modifies, and terminates multimedia sessions like IP telephony by registering systems and providing SIP addresses to check availability and invite others to sessions.
Video Teleconferencing (VTC) Technology at the National ...Videoguy
This document provides information about video teleconferencing (VTC) technology standards and practices at the National Institutes of Health (NIH). It discusses the business objectives for a standardized VTC architecture, including enabling firewall passage, remote management, standard naming conventions, dedicated telephony services, and compliance with ITU standards. It also describes the current VTC infrastructure and practices at NIH, including support for both internet protocol (IP) and integrated services digital network (ISDN) videoconferencing according to ITU recommendations H.323 and H.320. Finally, it lists the compliant specifications for VTC endpoints to integrate with the NIH infrastructure.
- Videoconferencing allows participants to see, hear and collaborate in real time over internet or telephone networks. It requires equipment like cameras, microphones, displays and codecs to compress and decompress audio/video data.
- Standards like H.320, H.323 and H.324 specify protocols for videoconferencing over ISDN, IP networks and POTS lines. Transport methods include ISDN, IP networks, cellular networks and POTS lines.
- Key components of videoconferencing systems are video/audio input/output devices, data transfer networks, and codecs. Formats like H.261, H.263, H.264 and audio standards G.711, G.722 are
This document summarizes key concepts and standards related to videoconferencing. It discusses audio teleconferencing, videoconferencing, and web conferencing. It defines videoconferencing as communication across long distances with video and audio using digital video transmission systems. It describes H.320 and H.323 as the main videoconferencing standards, with H.320 using ISDN lines and H.323 using packet-based networks like LANs. It also defines common videoconferencing terms like ISDN, codecs, bridges, full-duplex audio, and full-motion video.
The document provides an overview of videoconferencing basics, including what videoconferencing is, its benefits, how it is being used, common system components, and top manufacturers. It also discusses network connections, protocols, and different types of conferencing such as point-to-point, multipoint, webcasting, web conferencing, and audio conferencing.
The document summarizes INTEGRIS's video conferencing infrastructure and plans. It describes their transition from dedicated H.320 connections to an IP-based system using H.323 and H.264 standards across their WAN. Their current Polycom bridge supports 24 sites at 384kbps or 12 sites at 768kbps. Upgrading the bridge would allow mixing newer H.264 endpoints with older H.263 ones. Newer desktop and all-in-one systems offer improved video quality but require additional bandwidth or an upgrade to fully utilize the H.264 standard.
H.323 is the standard for multimedia conferences over IP networks. This document discusses implementing QoS solutions for H.323 video conferencing over an enterprise WAN with low-speed links. It provides guidance on capacity planning, determining bandwidth needs, classifying and prioritizing traffic. The recommended approach is to use DSCP values to classify voice as EF, video as AF41, and control traffic as AF31. Queuing mechanisms like LLQ and CBWFQ are suggested to provide minimum bandwidth guarantees for real-time traffic.
This document provides information about HEAnet's video conferencing service, which allows institutions to conduct video conferences over IP networks using H.323 standards. It describes the basic principles and elements of H.323 video conferencing. To use the service, an institution needs an H.323 compatible terminal and a connection to HEAnet's network. The document outlines how to register a terminal with HEAnet's gatekeeper to obtain a Global Dialing Scheme (GDS) number and make video calls within the HEAnet network or to external connections. Security considerations for opening necessary ports on an institution's firewall are also discussed.
This document provides an overview of multimedia services over IP networks and discusses two key protocols used: SIP and H.323. It describes the basics of SIP including session descriptions using SDP, message format, and session initiation. It also discusses SIP applications like IMS including requirements, protocols used, and architecture. For H.323, it outlines the network architecture including terminals, MCUs, gateways, and gatekeepers. It then describes the H.323 signaling protocols including RAS, H.225 call signaling, and H.245 call control.
This document provides an outline on voice over internet protocol (VoIP) covering topics such as how VoIP works, advantages of VoIP, types of codecs used for converting analog signals to digital data, signaling protocols like H.323 and SIP used for setting up and managing calls, and security considerations in VoIP like the use of SRTP for encryption. The document compares VoIP to traditional PSTN telephone networks and PBX systems, and explains that VoIP uses packet switching over the internet to make phone calls instead of dedicated circuits, allowing for upgrades with only increased bandwidth.
The document discusses videoconferencing standards defined by the ITU, including H.323, H.320, audio standards like G.711, video standards like H.261 and H.263, communication standards, and T.120 for data collaboration. It provides an overview of key standards, their components and functions, such as terminals, gatekeepers, and gateways in H.323. Interoperability between devices is enabled by standards for areas like frame structure, synchronization, control signaling, and encryption.
ISDN was initially developed to provide integrated digital services over telephone networks, offering advantages like higher data speeds and lower noise compared to traditional phone lines. However, the rise of technologies like ADSL have reduced ISDN's advantages. While ISDN remains well-suited for real-time communications like voice calls and video conferencing due to its circuit-switched synchronous connections, some companies incorrectly use backup ISDN lines in ways that expose their private networks to security risks from public telephone networks. The document recommends using ISDN's circuit-switched connections for applications like multi-location video conferencing while avoiding insecure backup configurations.
Video conferencing services allow multiple locations to communicate simultaneously through two-way video and audio transmissions. It differs from regular video calls in that it is designed for conferences between multiple locations rather than individuals. Video conferencing was first commercially deployed in the 1970s and has since seen significant adoption in business, education, and healthcare due to improvements in broadband infrastructure and video compression techniques. Modern video conferencing is readily available to the general public at reasonable costs using standards-based technologies.
2014 innovaphone different protocols for different thingsVOIP2DAY
The document discusses various protocols used for unified communications, including H.323, SIP, H.460.17, ICE, DTLS-SRTP, and WebRTC. It summarizes the purpose and functionality of each protocol, how they enable connections through firewalls and NAT, and their advantages and disadvantages. WebRTC in particular allows for real-time communication within a web browser without plugins using HTML5 and JavaScript, establishing direct peer-to-peer connections through techniques like STUN and ICE.
Video Conferencing : Fundamentals and ApplicationVideoguy
The document discusses video conferencing fundamentals and applications. It covers topics like modes of video conferencing, components, technologies, standards, protocols, bandwidth requirements, quality of service factors, challenges, and the eBaithak desktop video conferencing system developed at IIT Kharagpur.
IP Centric Conferencing IP Centric Conferencing IP Centric ...Videoguy
IP Centric Conferencing involves real-time voice, video, and data conferencing over IP networks, allowing multiple participants to communicate simultaneously. Key benefits include reduced costs, improved communications, and increased business competitiveness. Implementing IP conferencing requires choices around standards, network devices, bandwidth, quality of service, scalability, management, security, and legacy system interconnectivity. Proper planning is needed to ensure a successful IP conferencing solution.
The document outlines the key features and capabilities needed for an ideal global multimedia collaboration system. It discusses limitations of existing solutions like H.323, SIP, Skype and Access Grid. The ideal system would be service-oriented, support large conferences, integrate different clients and protocols, and leverage distributed computing resources through a messaging backbone like NaradaBrokering. It describes the architecture of the Global-MMCS which aims to meet these goals using open-source components and a distributed broker network for scalable media delivery.
The document discusses current collaboration technologies like videoconferencing, instant messaging, and VoIP and their limitations. It introduces the Global Multimedia Collaboration System (Global-MMCS) which aims to provide a more advanced and integrated collaboration platform. Global-MMCS will use a service-oriented architecture to host media processing and session control services in a scalable and distributed manner, supporting various conferencing styles from massive broadcasts to small private meetings.
The document outlines the key features and capabilities needed for an ideal global multimedia collaboration system. It discusses limitations of existing solutions like H.323, SIP, Skype and Access Grid. The ideal system would be service-oriented, support large conferences, integrate different clients and protocols, and leverage distributed computing resources through a messaging backbone like NaradaBrokering. It describes the architecture and components of the Global-MMCS being developed, including distributed media services, XGSP conference control, and an open-source community grid approach.
Direct routing allows organizations to connect their on-premises PBXes or SIP trunks to Microsoft Teams via certified session border controllers (SBCs). This provides a full enterprise calling experience within Teams. The document discusses what direct routing is and when to use it, how to plan direct routing implementations including SBC options, and how to configure direct routing in Teams through settings like voice routing policies and phone number translations. It aims to help organizations optimize their direct routing configurations to leverage existing telephony infrastructure with Microsoft Teams.
Prosody S from Aculab is a sophisticated telephony media processing software with fully integrated SIP stack. It is extensively used within contact centre and blue light solutions.
Voice over IP (VoIP) allows voice traffic to be carried over an IP data network at lower bandwidth than traditional telephone networks. It provides benefits such as lower communication costs, convergence of voice and data infrastructure, and new multimedia applications. However, VoIP also faces issues including delay, congestion, jitter, packet loss, bandwidth limitations, echo, interoperability between different systems, and ensuring scalability. The two main VoIP protocols are the Session Initiation Protocol (SIP) and H.323. VoIP adoption is growing due to the increasing use of IP networks, and it provides opportunities for lower telephone costs and innovative services. However, challenges remain regarding quality of service, interoperability, and developing carrier-grade
Similar to Cost Efficient H.320 Video Conferencing over ISDN including ... (20)
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.
Cost Efficient H.320 Video Conferencing over ISDN including ...
1. Cost Efficient H.320 Video Conferencing over ISDN including Multipoint access
Cost Efficient H.320 Video Conferencing
over ISDN including Multipoint Access
Overview:
The purpose of this paper is to explain in greater detail how to implement cost savings into H.320 compliant ISDN based Video Conferences,
including conferences with multiple participants. It describes the components and their functions that are needed to establish an H.320 Video
Conference as well as how to implement cost efficient alternatives.
It is assumed that the reader has a general knowledge of Video Conferencing systems and the standards involved. However, the following
technical papers are available to provide more information on these topics:
■ How do I choose a Video Conferencing system?
■ Video Conferencing Standards and Terminology.
■ H.323 Terminals, Gatekeepers, Gateways & MCUs.
■ Global Dialling Scheme (GDS) for Schools VideoConferencing.
■ H.323 Dial Plan and Service Codes used by Gatekeepers etc.
■ IP Ports and Protocols used by H.323 Devices.
■ H.221 Framing used in ISDN Conferences.
Which standard do you need, H.320 or H.323?
The first question that starts the process of identifying your Video Conferencing system is concerned with who and where are the people that
you want to conference with. It is a networking issue that determines how the participating endpoints are going to be connected and hence
which is the applicable standard that you should consider following. As indicated above, there are effectively two standards used in Video
Conferencing, H.320 or H.323. After you have decided which standard you want to adhere too, you can start looking at the platform,
performance and price equation.
Do you want to conference just within your organisation, or with suppliers or the world?
H.320 is the ITU's umbrella standard for Video Conferencing between endpoints connected over ISDN, whilst H.323 is the ITU's umbrella
standard for Video Conferencing between endpoints connected over an IP network. Whilst the long term prediction is for the world to use IP, if
you currently do not have an existing LAN or WAN to connect the participants, then it is likely that the easiest and most cost efficient means of
connecting all the endpoints is via ISDN, especially if they are in different countries. Hence you need H.320 compliant endpoints, or better still,
dual H.323 and H.320 compliant endpoints that will allow you to migrate from ISDN to IP as and when the network infrastructure is available.
H.320 Endpoints:
Historically, H.320 endpoints were large and expensive room based systems that had to be booked in advance. Operating in this way is
restrictive, causes time delays and looses the spontaneity and momentum of using Video Conferencing to solve problems. However, we now
have H.320 compliant Desktop systems, Group systems and ISDN VideoPhones that make Video Conferencing generally more
available to everybody in an organisation.
Integrated Services Digital Network, (ISDN):
2. Cost Efficient H.320 Video Conferencing over ISDN including Multipoint access
An ISDN connection has two possible interfaces; a BRI (Basic Rate Interface) or a PRI (Primary Rate Interface). The BRI consists of two
circuit-switched B-channels, each of 64kbps that are used for data and one D-channel of 16kbps that is used for network control. The BRI
physically consists of two pairs of twisted wire (transmit & receive) that are terminated by an NTU (Network Termination Unit) in the form of
an RJ-45 connector. The PRI is similar to the BRI, but with more channels and extra control bandwidth. In Europe, the PRI consists of up to
thirty 64kbps B-channels, that are used for data transmission up to 1920kbps and one 64kbps D-channel for network control. The PRI is
usually terminated by an NTU in the form of two BNC connectors; although some use an RJ-45. In North America, the PRI consists of 23 B-
channels of 64kbps for data, one D-channel of 64kbps for network control and an extra 8kbps for Framing.
With H.320 Video Conferencing systems, each 64kbps B-channel can either have its own ISDN number; share the same
number as its paired BRI channel or be set so that all BRI channels share the same ISDN number. Usually, a BRI has the same ISDN number
assigned to both of its B-channels; this is controlled by the providing Telco. With a 384kbps Video Conferencing system, the
ability to operate in 6B mode or inverse multiplex the six channels together and operate in a 384 BONDED mode is a function of the
equipment. Either way, it will have six ISDN numbers that may or may not be the same.
In 6B mode, the 384kbps channel can be regarded as consisting of six individual B-channels. When initiating a conference call to a system
that is to operate in 6B mode, the six ISDN numbers must all be entered and dialled in the exact sequence that the ISDN lines are connected
to the equipment.
In 384 BONDED mode, the 384kbps channel can be regarded as consisting of six 64kbps TimeSlots, with the first TimeSlot, being
structured exactly like that for a B-channel. When initiating a conference call to a system that is to operate in 384 BONDED mode, only
the first ISDN number is entered and dialled. The receiving system acknowledges the call on the first channel and replies with its remaining
five numbers in the correct sequence. It is therefore crucial that a system is setup to reflect all its ISDN numbers in the correct sequence as
this is the only way in which a calling system can determine what to dial. The correct numbers are the local numbers without area code.
Only when all six ISDN numbers are known can the dialling system initiate a 384 BONDED call.
There are advantages and disadvantages to operating in a 384 BONDED versus 6B channel mode; with the obvious advantage being
that it can carry an extra 8kbps of Video. The conference is also initiated quicker as all six channels are dialled together. With 6B, the
channels are dialled sequentially, hence with a long distance call, there is a possibility that the first channel will time-out before the conference
is established. The main disadvantage is that all channels must be available for 384 BONDED to work. There is no recovery mechanism.
If, for whatever reason, a line is dropped during a conference, then the call is terminated.
An important consideration is the effect and use of Gateways and Multipoint Control Units. Most of these only support
BONDED calls at 384kbps and do not work in 6B mode. In these situations, you have no option but to use 384 BONDED calls.
Dedicated Multipoint Control Units, (MCU):
In the past, most H.320 conferences would have been between just two participants as ISDN is essentially a point-to-point connection.
However, multipoint technology now makes it possible for groups of people to participate in a conference and share information. To hold a
multipoint conference over ISDN, participants must use either a dedicated Multipoint Control Unit (MCU) that connects and manages all the
ISDN lines, or an endpoint with an embedded H.320 multipoint capability such as the Polycom HDX 8004XLP or Emblaze-VCON xPoint.
The basic function of any H.320 MCU is to maintain the communications between all the participants in the conference. H.320 MCU's are
hardware based as they need to connect to all of the ISDN lines from each participant. For example, to manage a conference between four
H.320 systems, each at 384kbps (3xBRI), a dedicated H.320 MCU needs to connect the twelve BRI's. This is typically done as 24 x 64kbps
3. Cost Efficient H.320 Video Conferencing over ISDN including Multipoint access
channels within a Primary Rate Interface, (PRI).
MCU's are capable of operating in either Continuous Presence or Voice-Activated Switching mode. Continuous
Presence allows participants to see more than just who is speaking. The actual number of participants viewable in a Continuous Presence
conference is a function of the MCU used and maybe subject to network constraints.
In Voice-Activated Switching mode everybody sees the participant who is speaking. When somebody else speaks, the MCU switches the
video and audio to the new speaker... and so on throughout the duration of the conference.
In general, dedicated MCU's support simultaneous sessions, more participants, higher bitrates, more screen layout options and more features
than embedded MCU's found in some endpoints.
Endpoint with Embedded H.320 Multipoint:
An alternative to using a dedicated MCU for small conferences involving 3 or 6 participants is to equip one of the endpoints with an embedded
multipoint capability. Both the Polycom HDX 7002XLP and Emblaze-VCON xPoint have an embedded multipoint options that support
themself and the other sites in either a Voice-Activated or Continuous Presence session.
Furthermore, both of these systems have IP connectivity as standard, so when their ISDN connectivity option is used in conjunction with their
multipoint capability, they allow mixed-mode operation between both ISDN and IP networks.
4. Cost Efficient H.320 Video Conferencing over ISDN including Multipoint access
In a simplistic manner, they act like a Gateway bridging between the other ISDN and IP endpoints.
Using an MCU Bureau Service:
A dedicated hardware H.320 MCU can represent a large capital investment that maybe difficult to justify when initially starting to use
H.320 Video Conferencing. How many times a month are you going to conduct a 3 or more way Video Conference? This could be
difficult to quantify until you actually start conferencing, so how can you justify the high capital investment when you don't really know how
often it will be used? The answer is you can't, so use a Bureau Service instead.
By initially using a Bureau Service to provide and manage whenever you need to conduct a 3 or more way Video Conference, you can
quantify how often you exactly require this functionality. Over a period of time, you will receive usage and cost reports, so you will be able to
determine with a reasonable amount of certainty if you can actually justify the capital investment required to have you own MCU. And if you
can't, then you can continue to use the Bureau Service.
21st Century Video offers a Bureau Service, for more information, please email: Bureau Services.
Cheaper ISDN Call Charges:
The ISDN Call charges can represent a significant amount of the ongoing costs associated with using H.320 Video
Conferencing, especially if the endpoints are overseas. A National 64kbps ISDN call costs a similar amount to that of a standard
telephone call, so a 384kbps ISDN call is at least 6 times that of a telephone call to the same destination. However, overseas ISDN calls have
a higher tariff. An alternative is to use Least Cost Call Routing, (LCCR) that offers substantial savings over British
Telecommunication’s standard rate for both UK and International ISDN calls.
21st Century Video can enable LCCR, for more information, please email: Least Cost Call Routing.
Updated: 01 January 2010.