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.
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.
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.
This document provides a step-by-step guide for setting up SIP trunking between two KX-NS PBX systems to allow calls between them while displaying correct caller ID information. It outlines 26 programming steps to configure the PBXs, including assigning IP addresses and ports, enabling SIP registration, setting DDI, CLIP, dialing plans and more. Upon completion of the setup, calls can be made between extensions on either PBX and the correct caller ID will be displayed for the calling party. Revisions were made to the document on three dates to update pictures and explanations.
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 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.
Matrix Telecom Solutions: SETU VTEP - Fixed VoIP to T1/E1 PRI GatewayMatrix Comsec
Matrix SETU VTEP is a compact and dedicated gateway for VoIP to T1/E1 PRI network offering high-value communication experience to businesses of all size, Service Providers, Call Centers and simple but cost-effective solution for multi-location branch office communication. This intelligently designed gateway incorporates advanced features with multiple connectivity options to connect with a legacy communication system using T1/E1 or PRI signaling. SETU VTEP offers reliable and cost-effective solutions to the changing requirements of the business communication and offer customer value for money.
- The document provides an agenda and overview for a technical training on the SETU VTEP device, which is a VoIP-ISDN PRI gateway.
- It describes the device's interfaces, port configuration, hardware architecture, and LED indications for power, reset sequence, SIP trunk status, and T1/E1 port alarms.
- It also covers installation guidelines, applications of the device for VoIP access and PRI gateway functions, and configuration of incoming call routing based on calling number, called number, or DDI number.
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 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.
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.
This document provides a step-by-step guide for setting up SIP trunking between two KX-NS PBX systems to allow calls between them while displaying correct caller ID information. It outlines 26 programming steps to configure the PBXs, including assigning IP addresses and ports, enabling SIP registration, setting DDI, CLIP, dialing plans and more. Upon completion of the setup, calls can be made between extensions on either PBX and the correct caller ID will be displayed for the calling party. Revisions were made to the document on three dates to update pictures and explanations.
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 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.
Matrix Telecom Solutions: SETU VTEP - Fixed VoIP to T1/E1 PRI GatewayMatrix Comsec
Matrix SETU VTEP is a compact and dedicated gateway for VoIP to T1/E1 PRI network offering high-value communication experience to businesses of all size, Service Providers, Call Centers and simple but cost-effective solution for multi-location branch office communication. This intelligently designed gateway incorporates advanced features with multiple connectivity options to connect with a legacy communication system using T1/E1 or PRI signaling. SETU VTEP offers reliable and cost-effective solutions to the changing requirements of the business communication and offer customer value for money.
- The document provides an agenda and overview for a technical training on the SETU VTEP device, which is a VoIP-ISDN PRI gateway.
- It describes the device's interfaces, port configuration, hardware architecture, and LED indications for power, reset sequence, SIP trunk status, and T1/E1 port alarms.
- It also covers installation guidelines, applications of the device for VoIP access and PRI gateway functions, and configuration of incoming call routing based on calling number, called number, or DDI number.
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.
1. The document introduces Session Initiation Protocol (SIP), explaining that it is an application layer signaling protocol for initiating, modifying, and terminating multimedia communication sessions over IP such as voice and video calls.
2. It describes why SIP is used, including for conferencing, distance learning, video conferencing, instant messaging, and voice calls. It also outlines the main components of a SIP network including user agents, proxies, and redirects servers.
3. The document provides an overview of how SIP works by outlining the signaling process for registration, call setup and teardown, redirection, and media routing between user agents.
Matrix Telecom Solutions: SETU VGB - Fixed VoIP to GSM/3G-ISDN BRI GatewayMatrix Comsec
SETU VGB is an integrated gateway offering connectivity to IP, GSM/3G and ISDN BRI networks on a single platform. The gateway offers access to IP and GSM/3G networks for existing users of ISDN PBX. For an IP-PBX, it provides BRI and GSM/3G trunks connectivity. The gateway offers 8 VoIP channels, 4 GSM/3G SIMs and 2 ISDN BRI ports
VoIP security involves threats like denial of service attacks, eavesdropping, and quality of service issues. Best practices include using firewalls with application layer gateways or session border controllers, encrypting media and signaling, prioritizing bandwidth for VoIP, and restricting access to call managers through physical and logical security measures. NIST recommends logically separate networks, endpoint encryption, and avoiding vulnerabilities in softphones and wireless networks without encryption.
VoIP allows for transmitting voice calls over TCP/IP networks instead of traditional circuit-switched networks. It started gaining popularity in the mid-1990s but had drawbacks due to lack of broadband. VoIP offers unlimited distance, lower costs, and uses standards-based protocols like H.323, SIP, and MGCP. Tadiran deployed VoIP across multiple sites globally using Universal Gateways and IP phones.
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.
This document provides an overview of IP and VoIP fundamentals including:
- IP basics such as IP addressing schemes (IPv4 and IPv6), IP ranges, and private IP address ranges.
- VoIP concepts such as devices, codecs, and channels.
- SIP including messages, responses, and call flows for peer-to-peer and proxy calling.
- SIP trunks and their differences from VoIP channels.
- SIP extensions and configuring them on a server.
- VoIP port configuration including LAN, WAN, DNS, and STUN/port forwarding.
- Matrix products that support VoIP including their VoIP channel and trunk capabilities.
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 videoconferencing technologies and standards. It discusses H.323 as the dominant standard, describing its components like gatekeepers, terminals, and multimedia algorithms. It also covers conferencing versus broadcasting, networking considerations like switches versus hubs, and challenges with firewalls and network address translation.
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.
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.
The document discusses configuring VoIP and video calling on an ETERNITY NE system, including setting up network parameters, SIP trunks and extensions, registering Matrix IP phones as extensions, and making audio and video calls between SIP extensions using phones like SPARSH VP248, softphones, and the SPARSH M2S for video calls.
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.
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.
This document provides an overview and instructions for configuring and using a VoIP-FXO-FXS Gateway. It discusses the gateway's interfaces, hardware architecture, LED indicators, installation guidelines, applications, and programming options using a phone or PC. Settings covered include port configuration, incoming/outgoing call management, and advanced options. The gateway allows connecting analog phones to an IP network using SIP protocol and provides voice services over IP for SOHO and small/medium businesses.
The presentation is a compiled assembly from the SIP RFC' s, and original works of Alan Johnston and Henry Sinnreich . It contains Sip Detailed , Call flows , Architecture descriptions , SIP services , sip security , sip programming.
This document describes the ETERNITY IP-PBX system and its features. It is targeted at small to large businesses to provide telephony solutions. The ETERNITY system provides IP-PBX functionality with support for VoIP, analog, ISDN, and GSM/3G interfaces. It offers features such as seamless mobility, universal connectivity, and productivity and cost savings tools.
1. The document provides an overview of the Matrix ETERNITY integrated enterprise voice switch, which can support up to 1,344 user ports and provides interfaces for POTS, ISDN, GSM, VoIP, and other systems.
2. ETERNITY supports a variety of extension types including SLT, DKP, SIP, and ISDN terminals. It also interfaces with external devices like fax machines, door phones, and public address systems.
3. ETERNITY is available in different configurations including the LE, ME, GE, and PE models, which provide different capacities for user ports, trunks, and features. The LE can support the most users and trunks while the PE
Matrix ETERNITY is a family of IP-PBXs with Universal Connectivity and Seamless Mobility. The ETERNITY IP-PBX offers built-in gateway capability to connect nearly all telecom interfaces like FXS, FXO, ISDN BRI, ISDN PRI, T1/E1, GSM and 3G.
This document provides instructions for configuring VoIP settings on an ETERNITY hybrid IP-PBX, including LAN/WAN port configuration, MAC cloning, dynamic DNS, VoIP server domain, STUN, VLAN, VoIP port parameters, SIP extensions, SIP trunks, and Matrix extended phones. Key steps outlined include configuring the LAN and WAN ports, enabling MAC cloning to authenticate with the ISP, setting up dynamic DNS for dynamic public IPs, and configuring SIP extension and trunk parameters like authentication, codecs, and timers.
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.
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.
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.
1. The document introduces Session Initiation Protocol (SIP), explaining that it is an application layer signaling protocol for initiating, modifying, and terminating multimedia communication sessions over IP such as voice and video calls.
2. It describes why SIP is used, including for conferencing, distance learning, video conferencing, instant messaging, and voice calls. It also outlines the main components of a SIP network including user agents, proxies, and redirects servers.
3. The document provides an overview of how SIP works by outlining the signaling process for registration, call setup and teardown, redirection, and media routing between user agents.
Matrix Telecom Solutions: SETU VGB - Fixed VoIP to GSM/3G-ISDN BRI GatewayMatrix Comsec
SETU VGB is an integrated gateway offering connectivity to IP, GSM/3G and ISDN BRI networks on a single platform. The gateway offers access to IP and GSM/3G networks for existing users of ISDN PBX. For an IP-PBX, it provides BRI and GSM/3G trunks connectivity. The gateway offers 8 VoIP channels, 4 GSM/3G SIMs and 2 ISDN BRI ports
VoIP security involves threats like denial of service attacks, eavesdropping, and quality of service issues. Best practices include using firewalls with application layer gateways or session border controllers, encrypting media and signaling, prioritizing bandwidth for VoIP, and restricting access to call managers through physical and logical security measures. NIST recommends logically separate networks, endpoint encryption, and avoiding vulnerabilities in softphones and wireless networks without encryption.
VoIP allows for transmitting voice calls over TCP/IP networks instead of traditional circuit-switched networks. It started gaining popularity in the mid-1990s but had drawbacks due to lack of broadband. VoIP offers unlimited distance, lower costs, and uses standards-based protocols like H.323, SIP, and MGCP. Tadiran deployed VoIP across multiple sites globally using Universal Gateways and IP phones.
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.
This document provides an overview of IP and VoIP fundamentals including:
- IP basics such as IP addressing schemes (IPv4 and IPv6), IP ranges, and private IP address ranges.
- VoIP concepts such as devices, codecs, and channels.
- SIP including messages, responses, and call flows for peer-to-peer and proxy calling.
- SIP trunks and their differences from VoIP channels.
- SIP extensions and configuring them on a server.
- VoIP port configuration including LAN, WAN, DNS, and STUN/port forwarding.
- Matrix products that support VoIP including their VoIP channel and trunk capabilities.
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 videoconferencing technologies and standards. It discusses H.323 as the dominant standard, describing its components like gatekeepers, terminals, and multimedia algorithms. It also covers conferencing versus broadcasting, networking considerations like switches versus hubs, and challenges with firewalls and network address translation.
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.
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.
The document discusses configuring VoIP and video calling on an ETERNITY NE system, including setting up network parameters, SIP trunks and extensions, registering Matrix IP phones as extensions, and making audio and video calls between SIP extensions using phones like SPARSH VP248, softphones, and the SPARSH M2S for video calls.
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.
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.
This document provides an overview and instructions for configuring and using a VoIP-FXO-FXS Gateway. It discusses the gateway's interfaces, hardware architecture, LED indicators, installation guidelines, applications, and programming options using a phone or PC. Settings covered include port configuration, incoming/outgoing call management, and advanced options. The gateway allows connecting analog phones to an IP network using SIP protocol and provides voice services over IP for SOHO and small/medium businesses.
The presentation is a compiled assembly from the SIP RFC' s, and original works of Alan Johnston and Henry Sinnreich . It contains Sip Detailed , Call flows , Architecture descriptions , SIP services , sip security , sip programming.
This document describes the ETERNITY IP-PBX system and its features. It is targeted at small to large businesses to provide telephony solutions. The ETERNITY system provides IP-PBX functionality with support for VoIP, analog, ISDN, and GSM/3G interfaces. It offers features such as seamless mobility, universal connectivity, and productivity and cost savings tools.
1. The document provides an overview of the Matrix ETERNITY integrated enterprise voice switch, which can support up to 1,344 user ports and provides interfaces for POTS, ISDN, GSM, VoIP, and other systems.
2. ETERNITY supports a variety of extension types including SLT, DKP, SIP, and ISDN terminals. It also interfaces with external devices like fax machines, door phones, and public address systems.
3. ETERNITY is available in different configurations including the LE, ME, GE, and PE models, which provide different capacities for user ports, trunks, and features. The LE can support the most users and trunks while the PE
Matrix ETERNITY is a family of IP-PBXs with Universal Connectivity and Seamless Mobility. The ETERNITY IP-PBX offers built-in gateway capability to connect nearly all telecom interfaces like FXS, FXO, ISDN BRI, ISDN PRI, T1/E1, GSM and 3G.
This document provides instructions for configuring VoIP settings on an ETERNITY hybrid IP-PBX, including LAN/WAN port configuration, MAC cloning, dynamic DNS, VoIP server domain, STUN, VLAN, VoIP port parameters, SIP extensions, SIP trunks, and Matrix extended phones. Key steps outlined include configuring the LAN and WAN ports, enabling MAC cloning to authenticate with the ISP, setting up dynamic DNS for dynamic public IPs, and configuring SIP extension and trunk parameters like authentication, codecs, and timers.
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.
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.
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.
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.
- 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
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.
H.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network.
The H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences.
It is widely implemented by voice and videoconferencing equipment manufacturers, is used within various Internet real-time applications such as GnuGK and NetMeeting
It is widely deployed worldwide by service providers and enterprises for both voice and video services over IP networks.
It is a part of the ITU-T H.32x series of protocols, which also address multimedia communications over ISDN, the PSTN or SS7, and 3G
H.323 call signaling is based on the ITU-T Recommendation Q.931 protocol and is suited for transmitting calls across networks using a mixture of IP, PSTN, ISDN, and QSIG over ISDN.
Cost Efficient H.320 Video Conferencing over ISDN including ...Videoguy
- 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.
Cost Efficient H.320 Video Conferencing over ISDN including ...Videoguy
- The document discusses how to implement cost-efficient H.320 video conferencing over ISDN networks, including multipoint access. It describes the components needed for H.320 video conferencing and alternatives to reduce costs, such as using a dedicated multipoint control unit (MCU) or MCU bureau service to enable multipoint conferences. It also discusses using least cost call routing to reduce ISDN call charges, which can represent a significant ongoing cost.
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.
The document outlines an agenda for a site coordinator training on videoconferencing. It covers various topics including different videoconferencing technologies and standards like H.320 and H.323; components of an H.323 system like gatekeepers and MCUs; considerations for networking including switches vs hubs and duplex mismatch; and duties of a site coordinator.
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.
Collaborative conferencing options available to LTER Network ...Videoguy
Video teleconferencing (VTC) allows face-to-face meetings over the internet and is now commonly used by the LTER Network Office (LNO) and sites with success. Equipment options range from $1500-10000, with individual and small conference room options starting at $149 and $1999 respectively. The LNO has installed a $50,000 video bridge that can support multiple simultaneous meetings of up to 48 connections. Web conferencing is also available for lower bandwidth needs using shared tools through a web browser.
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.
Video conferencing allows two or more locations to interact simultaneously through two-way audio and video transmissions. It can improve work quality and productivity while reducing costs. The key components of a video conferencing system are cameras, displays, microphones, speakers, and broadband connectivity to exchange data between endpoints. Popular video conferencing equipment includes Polycom conference room units and personal units.
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.
VOIP allows IP networks to carry voice applications like telephony and conferencing. It uses protocols like SIP, H.323, and MGCP for signaling and codecs like G.711 and G.729 for compressing analog voice. Key VOIP components include IP phones, gateways, call agents, and MCUs. Signaling protocols establish and terminate sessions, with SIP and H.323 using a peer-to-peer model and MGCP using a client-server model. Considerations for VOIP include low jitter, latency under 150ms, minimal packet loss, and high availability to provide a reliable voice service over IP networks.
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 practical solutions for enabling H.323 video conferencing over limited bandwidth connections and through firewalls. It describes using quality of service (QoS) tagging and class-based weighted fair queueing (CBWFQ) on routers to prioritize H.323 traffic on connections with bandwidth limitations like T1 lines. It also discusses options for configuring firewalls and NAT to support H.323, including using a "smart" H.323 client configured for the public address, defining ports, or using an application proxy or firewall with H.323 fixup support.
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1. HEAnets’ Video Conferencing Service
1.-Principles of Video Conferencing
2.-Elements of a H.323 System
2.1.-Terminals
2.2.-Multipoint Control Units (MCUs)
2.3.-Gateways
2.4.-Gatekeepers
3.-What do Institutions need to get connected to HEAnet's video conferencing
service?
4.-Placing a H.323 terminal on the Network
5.-Security of H.323 service
6.-The Global Dialing Scheme (GDS)
7.-Registering your institutions H.323 terminal endpoint with HEAnet's gatekeeper
8.-Dialling test GDS Numbers
9.-Making a point to point video conferencing call
10.-Making a point to multipoint video conferencing call
11.-Continuous Presence point to multipoint conferences
12.-Private multipoint conferences and chairperson controlled conferences
13.-Dialling HEAnet's gateway (ISDN calls )
2. Principles of Video Conferencing
The principle of Video conferencing is that a meeting will take place between two or more sites
involving audio and video images. This results in time and cost savings between the
participants.
Using your Local Area Network (LAN) and your Wide Area Network (WAN) it is possible to send
these images across the public Internet. This results in further cost savings as most sites
already have a data network in place and there is no need for costly ISDN calls to take place.
The technology HEAnet are using to video conference uses the protocol H.323, this is based on
the more traditional ISDN standard H.320.
An excellent resource for H.323 related technology including video conferencing room design
may be found here.
Elements of a H.323 System
§ Terminals
§ Multipoint Control Units (MCUs)
§ Gateways
§ Gatekeepers
To use HEAnet's video conferencing service an institution only requires one of the four
elements above. Only endpoints are required. All other elements, MCU, gateway and
gatekeepers can be provided by HEAnet's service.
3. Terminals
To allow an institute to connect to HEAnet's multimedia services, one must have the required
devices that allow H.323 video conferencing. The basic systems that provide video and audio
images are known as H.323 terminals or endpoints.
HEAnet have tested the following terminals that have been proven to work on our systems.
§ Polycom Viewstation SP
§ Polycom ViaVideo
§ Vcon ViGO
§ Tandberg 550
§ Tandberg 880
§ Tandberg 10 00
There are numerous other endpoints that will also work with HEAnet's services. Generally if
the endpoint supports H.323 it should operate with HEAnet's equipment. If you wish to test if
an endpoint operates with HEAnet's services please contact multimedia
Different endpoints operate at different qualities. The basic way to differentiate between
different quality video conferences is on the speed that the terminal can connect to.
The average speed that H.323 terminals operate is at 384 kilo bits per second (kbps).
Video conference calls between 64 kbps and a 2Mbps are possible. HEAnet recommend that
calls should be made at either 384kbps or 768kbps.
4. Multipoint Control Units (MCUs)
MCU's allow one to participate in a point to multipoint conference. If more than two sites are
involved in a conference a MCU will be required.
Gateways
Gateways allow users to convert from older ISDN (H.320) video conferencing to IP video
conferencing (H.323) and vica versa. Gateways also allow GSM and traditional PSTN devices to
be involved in a conference.
5. Gatekeepers
Gatekeepers are used for admission control and address resolution of HEAnet's video
conferencing service.
What do Institutions need to get connected to HEAnet's
video conferencing service?
§ H.323 terminal.
§ Television. The h.323 terminal connects directly into a display unit such as a TV
system. A large TV system w ill enhance the quality of ones video conference. Please
ensure the sound system is audible to all participants in the room.
§ Lighting system. If a good quality camera system is used most general lighting
systems will be adequate for a video conference. However, direct light onto the local
participants faces will enhance the picture. Please also ensure that reflective surfaces
are kept to a minimum.
§ Furniture, Curtains, backdrop. A backdrop explaining where the video conference
room is located e.g. HEAnet Dublin, Ireland is a good idea in a large conference.
Curtains are used to limit visual reflection and also to dampen sounds. When choosing
furniture please ensure that all participants may be viewed though the camera of the
h.323 terminal.
§ LAN. If using h.323 conferences please ensure that there is no contention on your
network. Packet losses of 5% or more will render your conference unusable. Where
possible please use a switched ethernet design of 10/100 Mbps.
§ Wide area link to HEAnet. To use HEAnet's video conferencing service one must
have a WAN link to HEAnet. For good quality conferences HEAnet recommend at least
a 2Mbps connection.
6. Placing a H.323 terminal on the Network
Once the endpoint is obtained, one must place it on the premises Local Area Network. utting a
H.323 endpoint on the LAN involves much the same methods as placing a PC on a LAN. A
H.323 endpoint is essentially a PC with a camera and audio features. Therefore all of the
standard trouble shooting procedures should be used to ensure that the endpoint could
connect to the "outside world". Such procedures such as giving the device an IP address,
gateway address and so on are outside the scope of this document.
HEAnet recommend that if possible the video conference equipment is either placed on a
separate LAN or a Virtual one (VLAN) than the institutes main network, this will allow the
institutions to control both the security and the performance of the video conference better.
However, in saying that there should be no issues in placing the endpoint on your standard
LAN.
Once the standard settings are placed on the videoconference endpoint (IP addresses, subnet
mask, gateway etc.) it should be possible for one to connect out to the rest of the world and
utilise HEAnet's video conferencing services.
First of all check that one may make an IP video conference call to another endpoint. Dial the
following IP address 193.1.31.215. This should make a call to HEAnet's testing device that is
on auto answer.
If one can connect to this endpoint it is now possible to have a video conference call.
Once your H.323 terminal can reach the outside world, three easy steps are required to use
HEAnet's video conferencing service.
1. Ensure that all relevant services are available through your Internet security policy.
2. Get a GDS number.
3. Register with HEAnet's gatekeeper.
7. Security of H.323 service
Prior to using HEAnet's video conferencing service, there are a number of security issues
involved in allowing H.323 conferencing.
The following ports are a used in H.323 conferencing.
Function Port Type
HTTP Interface 80 TCP
Gatekeeper discovery 1718 TCP
Gatekeeper RAS 1719 TCP
Q.931 Call Setup 1720 TCP
Audio Call Control 1731 TCP
H.245 Control Channel 1024-65535 TCP
RTP ( Video / Audio ) 1024-65535 UDP
H.235 secure signaling 1300 TCP
T.120 1503 TCP
Because H.323 uses dynamically assigned ports it is difficult to control.
Some solutions to overcome this are as follows.
1. Open the ports listed above on your institutions firewall.
2. Use H.323 aware firewalls, e.g. Cisco Secure IS, Cisco PIX, Checkpoint FW-1, Netscreen.
3. Use a separate network for H.323 (H.323 DMZ).
8. The Global Dialing Scheme (GDS)
To use HEAnet's services one must first understand the Global Dialing Scheme (GDS). The
GDS is a numerical dialing plan that allows H.323 endpoints to dial to remote sites and
services. The GDS uses the same format as telephone numbers (E164) numbers, therefore a
call to the US means that one must dial the prefix 001, UK is 0044 and so on. More details of
the GDS may be found here.
As may be seen from the diagram above there are various numbering levels involved in the
GDS. Each institution has a three-digit number relating to their domain. The following list
gives the details of HEAnet's customers and Institutes of Technologies GDS numbers.
To allow one to use the GDS, one must register with HEAnet's gatekeeper. To register with
HEAnet's gatekeeper follow the points below.
If your institution is not on the list above and would like to subscribe t o the GDS, please
contact multimedia@heanet.ie
Each institution will be in control of their own GDS number, therefore should TCD who have a
GDS zone number of 121 decide to allocate numebrs, they can do so as follows.
Extension 1. 0035301121 001
Extension 2. 0035301121 002
Extension 3. 0035301121 003 etc………
The rest of the institutes will be the same; Dundalk 0035301114001 and so on.
Please note that one must use full-length numbers when dialing the GDS.
9. Registering your institutions H.323 terminal endpoint with
HEAnet's gatekeeper
1. In your videoconferencing endpoint unit, navigate to the H.323 settings menu.
2. Enter your email address as your endpoint's H.323 Name. This gives the gatekeeper
administrators some idea of who is registering and from where.
3. Assign an E.164 number (also called H.323 extension) in the following manner: 00 35301 +
your institute GDS number + your extension number. *
If, for example, your institutions GDS number is 0035301112 and your extension number is
002 (defined by each institutions IT department) your H.323 extension number would be
0035301112002.
4. Your endpoint's system may need to be rebooted.
5. Once it's running again, navigate back to the H.323 setting and select the gatekeeper
menu.
6. Enter the following IP address: 193.1.31.194
7. Again, your endpoint may need to be rebooted.
8. Congratulations. You should now be registered with HEAnet's Gatekeeper.
You can call the HEAnet Videoconferencing Service for additional information by calling (01)
6609040 or sending an email to multimedia@heanet.ie
The Irish national Gatekeeper is
193.1.31.194
Dialling test GDS Numbers
Once the three steps.
1. Ensure that all relevant services are available through your Internet security policy.
2. Get a GDS number.
3. Register with HEAnet's gatekeeper
are completed, one can now dial a worldwide GDS number.
To dial a GDS number ensure that you are dialling out through the H.323 system on your
terminal (called an IP call on some terminals). Then simply enter your GDS number and dial!
A test GDS number to use is 0035301101006 (in HEAnet's) office.Another test GDS number
is "copy bird" in the University of North Carolina in the USA. Dial 00112971216 to connect
to "copybird" and hear and see your own image.
Once you have confirmed that you can dial the above GDS numbers you can now dial any GDS
number and use HEAnet's videoconferencing services.
HEAnet encourage people to advertise their GDS numbers as publicly as possible, e.g. web
sites and email signatures. Only with awareness within the community will the system be
successful.
10. Making a point to point video conferencing call
To make a point to point (only two locations involved) videoconference, one simply needs to
know the GDS number of the remote location.
Once the GDS number is known, simply enter the GDS number in your destination field and
dial. On some systems, in particular Tandberg devices one must ensure that you have the
Network Profile set to LAN or IP call.
It is also possible to make a IP video conference call by entering in the IP address of the far
site, however this is not a scalable solution and to ensure that H.323 systems can expand in
the future, it is recommended that GDS numbers are used.
Making a point to multipoint video conferencing call
As already mentioned in this documentation to set-up a call with three or more destinations
involved will require an MCU device. HEAnet currently have a high specification Radvision MCU
that will allow upto 70 locations participate in a conference. Setting up a multipoint conference
involves one dialling a GDS number. At this point please ensure that you can dial GDS
numbers by testing to these numbers.
When thinking of multipoint conferences try to think of dialling a "virtual room" where
everyone meets. This "virtual room" therefore must have a GDS number to allow one to ring
it. Each member of HEAnet has its own "virtual room" based on its GDS number. For example
Trinity College Dublin has the GDS extension number of 122.
To dial TCD's "virtual room" simply dial the GDS number 003530110062122 To dial your own
institutions "virtual room" replace the 122 numbers with your own institutions GDS extension.
When organising a multipoint conference, one must decide in whose "virtual room" all parties
are to meet. Once this is decided in advance all parties dial that institutions "room". For
example if DCU, UCD, TCD and IT Carlow decide to meet in Trinity's "virtual room", at the
designated time all four institutions dial the number 003530110062122.
The "virtual room" number 003530110062122 is designed as a conference for terminals
connecting at a speed of 384kbps. Other connection speeds are available. For example if
Trinity wanted to set-up a conference at a lower speed e.g. 128kbps, the Trinity "virtual room"
for a 128kbps call is 003530110063122.
The following speeds are available for HEAnet members.
003530110062122 384kbps conference.
003530110063122 128kbps conference.
003530110064122 768kbps conference.
Please note that in all the GDS numbers above one should replace the 122 number with your
own individual Institution GDS number.
All of the multipoint conferences mentioned above are using a technology known as, voice
switched conferencing. This allows members of a multipoint conference to view the person
who is currently talking. For most meetings this type of conference is the most suitable.
However it is also possible to have a second type of conference known as continuous
presence.
11. Continuous Presence point to multipoint conferences
In voice switched conferences just one image at a time is displayed on the screen. However
with continuous presence it is possible to view four locations at once.
Continuous presence conference Voice Switched conference
To set-up a Continuous Presence conference simply dial the following GDS numbers.
003530110074122 384kbps conference.
003530110072122 128kbps conference.
Again the example above is for Trinity College Dublin's "virtual room". Please replace the 122
number with your own individual Institution GDS number.
12. Private multipoint conferences and chairperson c ontrolled
conferences
To set-up a password protected multipoint conference or chairperson controlled conference
one must fill out the following details and send the results to noc or multimedia in HEAnet.
Name of Conference: (e.g. TCD_7_4_2003)
Administrator of Conference
Name:
Institution:
Email:
Phone number:
GDS zone:
(Details of this may be found here)
Conference Details
Date of conference:
Time of conference (GMT time):
Number of locations in conference:
Required speed of conference: (e.g. 128kbps, 384kps, 768kbps)
Continuous Presence conference: YES/NO
(Please note Continuous Presence is only available ate 128kbps or 384kps)
Please fill out the following if Chairperson controlled conference is required.
Chairperson of Conference
Name:
Institution:
Email:
Phone number:
IP Address of desktop where control of conference will take place
Dialling HEAnet's gateway (ISDN calls)
Traditionally video conferencing has taken place over costly ISDN lines. HEAnet are using IP
calls H.323 for video conferencing.
To make legacy ISDN calls you can dial the HEAnet gateway and then be routed through to
the IP network.
To dial into our Gateway simply dial 01 4490889 (if within the Republic of Ireland). After
approximately 6 rings the gateway will answer. You will then be asked to dial a GDS number
followed by the # sign.
Some of the older video conferencing units do not have the ability to dial number strings when
the gateway answers. If this is the case it may be possible under the circumstances for
HEAnet to call your remote site. Please contact multimedia@heanet.ie for further details