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.
H.323 is a standard for multimedia communications over packet-based networks. It defines protocols for real-time audio, video and data communications between endpoints such as terminals, gateways and multipoint control units. As an umbrella standard, H.323 references other protocols for functions like call signaling, bandwidth negotiation and transmission of audio and video data. H.323 provides scalable and flexible multimedia communication capabilities and has been widely adopted for voice and video conferencing over both internet and private networks.
Internet protocol (VoIP) is the technology of digitizing sound, compressing it, breaking it up into data packets, and sending it over an IP network.The conventional technique used for sending voice is PSTN (public switched telephone network) . As data traffic has higher speed than telephone traffic, so what we do most of the time we prefer to send voice over data networks. Voice over internet protocol (VoIP) is a method of telephone communication over a data network.
VoIP, or Voice over Internet Protocol, is a technology that allows users to make voice calls using an Internet connection instead of a regular phone line. It works by converting voice signals into digital data packets that travel over the Internet and are then reconstructed at the other end. There are several VoIP protocols used and many applications that employ VoIP, including Skype. VoIP offers advantages over traditional phone service like lower costs, additional features included for free, and the ability to make calls from any Internet-connected device.
This document discusses VoIP (Voice over Internet Protocol) technologies. It begins by defining VoIP and how it allows phone calls to be made over the internet instead of traditional telephone networks. It then explores enterprise VoIP systems, hosted VoIP, VoIP phones, and the differences between circuit switching used in PSTN networks and packet switching used in VoIP. Challenges of VoIP like latency, jitter and packet loss are outlined, as well as advantages such as lower costs, flexibility and portability. Popular VoIP service providers like Google Voice and Skype are compared.
A complete power point presentation to know how Public Switching Telephone Network works. Useful for those in the working field or for the ones who want to know more or submitting any project report..
VoIP allows users to make phone calls using an Internet connection rather than a traditional phone line. It works by converting the voice signal from analog to digital, breaking it into packets, sending it over IP, reassembling it at the destination, and converting it back to analog. VoIP has advantages like low cost and portability but disadvantages like quality issues during power outages or network instability. Major challenges include addressing latency, echo, jitter, connection problems through firewalls and NAT, and overall reliability.
Integrated Services Digital Network (ISDN) is a set of communication protocols that provides digital transmission of voice, video, and data over telephone lines or normal telephone cables. ISDN was developed in the 1970s and provides end-to-end digital connectivity over digital media. ISDN services include bearer services to transfer information between networks, teleservices to allow networks to process content, and supplementary services that provide additional functionality.
This document summarizes circuit switching and packet switching techniques in communications networks. It discusses how circuit switching establishes a dedicated physical path between communicating nodes but is inefficient for bursty traffic. Packet switching breaks messages into packets that are transmitted over shared links, improving efficiency. Key aspects covered include virtual circuits, datagrams, packet switching advantages, X.25 standards, and how Frame Relay improved on X.25 by reducing overhead.
H.323 is a standard for multimedia communications over packet-based networks. It defines protocols for real-time audio, video and data communications between endpoints such as terminals, gateways and multipoint control units. As an umbrella standard, H.323 references other protocols for functions like call signaling, bandwidth negotiation and transmission of audio and video data. H.323 provides scalable and flexible multimedia communication capabilities and has been widely adopted for voice and video conferencing over both internet and private networks.
Internet protocol (VoIP) is the technology of digitizing sound, compressing it, breaking it up into data packets, and sending it over an IP network.The conventional technique used for sending voice is PSTN (public switched telephone network) . As data traffic has higher speed than telephone traffic, so what we do most of the time we prefer to send voice over data networks. Voice over internet protocol (VoIP) is a method of telephone communication over a data network.
VoIP, or Voice over Internet Protocol, is a technology that allows users to make voice calls using an Internet connection instead of a regular phone line. It works by converting voice signals into digital data packets that travel over the Internet and are then reconstructed at the other end. There are several VoIP protocols used and many applications that employ VoIP, including Skype. VoIP offers advantages over traditional phone service like lower costs, additional features included for free, and the ability to make calls from any Internet-connected device.
This document discusses VoIP (Voice over Internet Protocol) technologies. It begins by defining VoIP and how it allows phone calls to be made over the internet instead of traditional telephone networks. It then explores enterprise VoIP systems, hosted VoIP, VoIP phones, and the differences between circuit switching used in PSTN networks and packet switching used in VoIP. Challenges of VoIP like latency, jitter and packet loss are outlined, as well as advantages such as lower costs, flexibility and portability. Popular VoIP service providers like Google Voice and Skype are compared.
A complete power point presentation to know how Public Switching Telephone Network works. Useful for those in the working field or for the ones who want to know more or submitting any project report..
VoIP allows users to make phone calls using an Internet connection rather than a traditional phone line. It works by converting the voice signal from analog to digital, breaking it into packets, sending it over IP, reassembling it at the destination, and converting it back to analog. VoIP has advantages like low cost and portability but disadvantages like quality issues during power outages or network instability. Major challenges include addressing latency, echo, jitter, connection problems through firewalls and NAT, and overall reliability.
Integrated Services Digital Network (ISDN) is a set of communication protocols that provides digital transmission of voice, video, and data over telephone lines or normal telephone cables. ISDN was developed in the 1970s and provides end-to-end digital connectivity over digital media. ISDN services include bearer services to transfer information between networks, teleservices to allow networks to process content, and supplementary services that provide additional functionality.
This document summarizes circuit switching and packet switching techniques in communications networks. It discusses how circuit switching establishes a dedicated physical path between communicating nodes but is inefficient for bursty traffic. Packet switching breaks messages into packets that are transmitted over shared links, improving efficiency. Key aspects covered include virtual circuits, datagrams, packet switching advantages, X.25 standards, and how Frame Relay improved on X.25 by reducing overhead.
Direct sequence spread spectrum (DSSS) spreads data over a wide frequency band by combining the data with a redundant bit sequence called a chipping code. It can transmit at 1, 2, 5.5, and 11 Mbps using different encoding and modulation schemes. Barker coding maps each data bit to an 11-bit sequence for 1-2 Mbps using DBPSK or DQPSK. Complementary code keying maps groups of data bits to unique 8-bit sequences for 5.5-11 Mbps using DQPSK phase shifts. DSSS occupies a 22 MHz band but can tolerate some interference due to its processing gain from spreading the signal.
VOICE OVER INTERNET PROTOCOL (VOIP) allows users to make phone calls using an Internet connection rather than a traditional phone line. VOIP compresses voice data, converts it to digital signals in IP packets, and transports them over the Internet or data networks. This provides economic benefits compared to traditional phone networks since the same infrastructure can be used for both data and voice. While VOIP provides advantages like low cost and ability to communicate anywhere, it faces challenges around voice delays and compatibility with existing phone networks. Key components of a VOIP system include clients, servers, and gateways to connect VOIP and traditional phone networks.
The document discusses the evolution of mobile communications technology from 1G to 5G standards. It provides details on the key technologies, features, and limitations of each generation. 1G systems used analog signals for voice only, while 2G introduced digital networks. 3G enabled broadband data and multimedia. 4G aimed for ultra-broadband speeds up to 1Gbps. 5G is expected to offer wireless internet access with almost no limitations at speeds over 1Gbps. Each new standard aimed to improve on the capabilities and speeds of prior generations.
The document discusses IP-telephony, which routes voice calls over internet or IP-based networks by transmitting data via packet switching. It describes the types of IP-telephony networks and devices, requirements, components, how VOIP works, advantages like lower costs and mobility, disadvantages like firewalls and reliability issues, and concludes by listing references for further information.
This document provides an overview of Voice over IP (VoIP) technology. It discusses how VoIP works by digitizing and transmitting voice signals over the internet using IP packets. It describes common VoIP protocols like H.323 and SIP. The advantages of VoIP include lower costs, flexibility, and the ability to make calls from any internet connection. Disadvantages include reliance on internet access and potential quality issues during network congestion. The document provides details on how to implement VoIP securely and protect against risks.
LTE (Long Term Evolution) is a 4G wireless technology designed to support higher data speeds and capacities. It uses OFDMA for the downlink and SC-FDMA for the uplink. LTE supports MIMO to increase data rates through multiple antennas. The LTE network architecture consists of the eNodeB base stations, Mobility Management Entity (MME) for control plane functions, Serving Gateway (SGW) for user plane functions, and Packet Data Network Gateway (PGW) connecting to external networks. Voice can be supported in LTE through Circuit Switched Fallback (CSFB) to legacy networks or using Voice over LTE (VoLTE) with IP Multimedia Subsystem (IMS
Voice over Internet Protocol (VoIP) is a technology that allows users to make voice calls using a broadband Internet connection instead of a regular phone line. VoIP converts voice signals from phone calls into digital data packets that travel over the Internet or a private network using protocols like SIP. This allows for phone calls between computers or VoIP-enabled phones and traditional phones at low cost. Some key requirements for VoIP include software for voice processing, call signaling, and packet processing, as well as hardware like IP phones and gateways to connect to the public switched telephone network. VoIP can be used for calls over the public Internet, between offices on a private network, or with an IP PBX for a business phone system. Advantages
Broadband-ISDN (B-ISDN) is an extension of ISDN that provides broadband capabilities over digital networks. B-ISDN uses asynchronous transfer mode (ATM) and supports transmission speeds greater than 1.544 Mbps. It provides fully integrated services including high-speed data, audio, and full-motion video. The goal of B-ISDN is to achieve complete integration of services from low-bit rate bursty signals to high-bit rate continuous real-time signals.
Overview of VoIP (Voice over IP) and FoIP (Fax over IP) technologies like Session Initiation Protocol and H.323.
Even though voice over IP (VoIP) was hailed as a technological innovation, the idea to transport real-time traffic over TCP/IP networks was not new back in the 1990s when VoIP started being deployed in networks. Chapter 2.5 of the venerable RFC793 (TCP) shows both data oriented application traffic as well as voice being transported over IP based networks.
Nevertheless, VoIP puts high demands on signal and protocol processing capabilities so it became possible at reasonable costs only in the 1990s.
VoIP can be roughly split into two main functions. Signaling protocols like SIP (Session Initiation Protocol), H.323 and MGCP/H.248 are used to establish a conference session and the data path for transporting real-time voice data packets. SIP has largely supplanted H.323 in recent years to its simpler structure and packet sequences. MGCP and H.248 are mostly used in carrier backbone networks.
Protocols like RTP (Real Time Protocol) transport voice packets and provide the necessary information for receivers to equalize packet flow variations to provide a smooth playback of the original voice signal.
Voice codecs are one of the core functions of the data path. Voice compression reduces the bandwidth required to transport voice over an IP based network. Compression may be less of a concern in local area networks with gigabit speeds, on slower links like 3G (UMTS, LTE) it still makes a lot of sense.
The algorithms used in different codecs make use of various characteristics of the characteristics of human speech recognition. Redundant information is removed from the signals thus slightly reducing the quality, but greatly reducing the required bandwidth.
In VoIP networks, the echo problem is typically compounded by the increased delay incurred by packetization of voice signals. To counteract the echo problem, VoIP gear (hard phones, soft phones, gateways) include echo cancelers to remove echo signals from the transmit signal.
To transport facsimile over an IP based network, even more technology is needed. Facsimile protocols are very susceptible to delay and delay variation and thus need more compensation algorithms. Protocols like T.38 terminate facsimile protocols like T.30 (analog facsimile) and transport the fax images as digitized pictures over IP based networks.
The document discusses GPON (Gigabit Passive Optical Network) technology and implementation models. It provides information on:
- GPON standards and components like the OLT, ONU, and splitters
- Implementation models for retail/residential, enterprise/HRB, and mobile backhaul networks
- Considerations for ODN design and link budget calculations for different splitting scenarios
- Capabilities of OLTs, ONUs, and ONTs including interfaces, services supported, and functionalities
- Examples of residential ODN installation and network architectures for different use cases
The document provides an overview of the public switched telephone network (PSTN). It discusses that the PSTN is the interconnected telephone system that uses copper wires to make circuit-switched calls. It then covers the evolution of the PSTN from its invention in 1876 to present digital switches, the use of bandwidth allocation and numbering schemes, and call setup which involves signaling and switching systems to route calls.
Wireless communication theodore rappaportDaud Khan
The document repeatedly lists the website "www.vsofts.net" and the word "oldroad" without any other context or information provided. It is not possible to determine the essential meaning or purpose of the document from the limited information given.
This document defines and compares two types of wireless local loop (WLL) technologies: Local Multipoint Distribution Service (LMDS) and Multichannel Multipoint Distribution Service (MMDS). LMDS operates above 20 GHz and provides high-speed broadband, while MMDS operates between 2.1-2.7 GHz and provides lower bandwidth but stronger signals over longer distances. Both can provide voice, data and video services as alternatives to wired local loops. Key differences are that LMDS supports higher data rates over shorter ranges while MMDS has a larger cell size and is more suitable for large networks.
Signaling System #7 (SS7) is a telecommunications protocol that defines high-speed circuit switching for telephone calls and uses out-of-band signaling between service switching points, signal transfer points, and service control points. It has advantages like separation of control information onto logically separate paths, message-oriented call information exchange, and ability of a single signaling channel to carry information about multiple trunks. The SS7 architecture includes service switching points, service control points, and signal transfer points that communicate using protocols like ISUP, TCAP, and SCCP.
- GPRS is an upgrade to GSM that allows packet-based data services and efficient use of network bandwidth. It provides higher data rates than GSM and constant connectivity.
- The GPRS network architecture introduces new network elements like the SGSN and GGSN to route data packets. The SGSN manages packet data in its service area while the GGSN connects the GPRS network to external packet networks.
- Session management in GPRS includes establishing PDP contexts for data transfer sessions and location management tracks the routing area of mobile devices through routing area updates.
This document provides an overview of Bluetooth technology, including its history, specifications, networks, layers, applications, and issues. Bluetooth was developed in the late 1990s to facilitate short-range wireless connectivity between devices. It uses radio waves and frequency hopping to transmit data between devices within a personal area network. Common applications of Bluetooth technology include connecting headphones, printers, and automobiles. While scalability and throughput are limitations, Bluetooth provides a simple, inexpensive way to connect electronic devices without wires.
Contents:
Data Traffic
Congestion
Congestion Control
Quality of Service
Techniques to improve QOS
How QOS is implemented within the Internet
References..
The document provides an overview of LTE (Long Term Evolution) network architecture and transmission schemes. It describes the simplified LTE network elements including eNB, MME, S-GW and P-GW. It explains the downlink transmission scheme using OFDMA and reference signal structure. It also covers uplink transmission using SC-FDMA, control and data channels as well as frame structure in both FDD and TDD modes.
This document summarizes information about video conferencing using H.323 standards. It discusses the H.323 protocols for audio and video, requirements for networks supporting H.323, and tools for diagnosing problems. It also provides an overview of ViDe.Net, an international virtual network providing video conferencing, and future directions for middleware to support directories and management of video users.
VoIP allows users to make phone calls using an Internet connection instead of a traditional phone line. It works by converting voice signals to digital data that is transmitted in packets over the Internet. A VoIP network uses protocols like SIP and RTP to setup calls and transmit voice data. Components include VoIP protocols, gateways to interface with the PSTN, and codecs to compress voice signals. Businesses are attracted to VoIP as it can help reduce costs while improving utilization of bandwidth and network management. However, security risks like hacking and eavesdropping exist since VoIP uses the public Internet.
Direct sequence spread spectrum (DSSS) spreads data over a wide frequency band by combining the data with a redundant bit sequence called a chipping code. It can transmit at 1, 2, 5.5, and 11 Mbps using different encoding and modulation schemes. Barker coding maps each data bit to an 11-bit sequence for 1-2 Mbps using DBPSK or DQPSK. Complementary code keying maps groups of data bits to unique 8-bit sequences for 5.5-11 Mbps using DQPSK phase shifts. DSSS occupies a 22 MHz band but can tolerate some interference due to its processing gain from spreading the signal.
VOICE OVER INTERNET PROTOCOL (VOIP) allows users to make phone calls using an Internet connection rather than a traditional phone line. VOIP compresses voice data, converts it to digital signals in IP packets, and transports them over the Internet or data networks. This provides economic benefits compared to traditional phone networks since the same infrastructure can be used for both data and voice. While VOIP provides advantages like low cost and ability to communicate anywhere, it faces challenges around voice delays and compatibility with existing phone networks. Key components of a VOIP system include clients, servers, and gateways to connect VOIP and traditional phone networks.
The document discusses the evolution of mobile communications technology from 1G to 5G standards. It provides details on the key technologies, features, and limitations of each generation. 1G systems used analog signals for voice only, while 2G introduced digital networks. 3G enabled broadband data and multimedia. 4G aimed for ultra-broadband speeds up to 1Gbps. 5G is expected to offer wireless internet access with almost no limitations at speeds over 1Gbps. Each new standard aimed to improve on the capabilities and speeds of prior generations.
The document discusses IP-telephony, which routes voice calls over internet or IP-based networks by transmitting data via packet switching. It describes the types of IP-telephony networks and devices, requirements, components, how VOIP works, advantages like lower costs and mobility, disadvantages like firewalls and reliability issues, and concludes by listing references for further information.
This document provides an overview of Voice over IP (VoIP) technology. It discusses how VoIP works by digitizing and transmitting voice signals over the internet using IP packets. It describes common VoIP protocols like H.323 and SIP. The advantages of VoIP include lower costs, flexibility, and the ability to make calls from any internet connection. Disadvantages include reliance on internet access and potential quality issues during network congestion. The document provides details on how to implement VoIP securely and protect against risks.
LTE (Long Term Evolution) is a 4G wireless technology designed to support higher data speeds and capacities. It uses OFDMA for the downlink and SC-FDMA for the uplink. LTE supports MIMO to increase data rates through multiple antennas. The LTE network architecture consists of the eNodeB base stations, Mobility Management Entity (MME) for control plane functions, Serving Gateway (SGW) for user plane functions, and Packet Data Network Gateway (PGW) connecting to external networks. Voice can be supported in LTE through Circuit Switched Fallback (CSFB) to legacy networks or using Voice over LTE (VoLTE) with IP Multimedia Subsystem (IMS
Voice over Internet Protocol (VoIP) is a technology that allows users to make voice calls using a broadband Internet connection instead of a regular phone line. VoIP converts voice signals from phone calls into digital data packets that travel over the Internet or a private network using protocols like SIP. This allows for phone calls between computers or VoIP-enabled phones and traditional phones at low cost. Some key requirements for VoIP include software for voice processing, call signaling, and packet processing, as well as hardware like IP phones and gateways to connect to the public switched telephone network. VoIP can be used for calls over the public Internet, between offices on a private network, or with an IP PBX for a business phone system. Advantages
Broadband-ISDN (B-ISDN) is an extension of ISDN that provides broadband capabilities over digital networks. B-ISDN uses asynchronous transfer mode (ATM) and supports transmission speeds greater than 1.544 Mbps. It provides fully integrated services including high-speed data, audio, and full-motion video. The goal of B-ISDN is to achieve complete integration of services from low-bit rate bursty signals to high-bit rate continuous real-time signals.
Overview of VoIP (Voice over IP) and FoIP (Fax over IP) technologies like Session Initiation Protocol and H.323.
Even though voice over IP (VoIP) was hailed as a technological innovation, the idea to transport real-time traffic over TCP/IP networks was not new back in the 1990s when VoIP started being deployed in networks. Chapter 2.5 of the venerable RFC793 (TCP) shows both data oriented application traffic as well as voice being transported over IP based networks.
Nevertheless, VoIP puts high demands on signal and protocol processing capabilities so it became possible at reasonable costs only in the 1990s.
VoIP can be roughly split into two main functions. Signaling protocols like SIP (Session Initiation Protocol), H.323 and MGCP/H.248 are used to establish a conference session and the data path for transporting real-time voice data packets. SIP has largely supplanted H.323 in recent years to its simpler structure and packet sequences. MGCP and H.248 are mostly used in carrier backbone networks.
Protocols like RTP (Real Time Protocol) transport voice packets and provide the necessary information for receivers to equalize packet flow variations to provide a smooth playback of the original voice signal.
Voice codecs are one of the core functions of the data path. Voice compression reduces the bandwidth required to transport voice over an IP based network. Compression may be less of a concern in local area networks with gigabit speeds, on slower links like 3G (UMTS, LTE) it still makes a lot of sense.
The algorithms used in different codecs make use of various characteristics of the characteristics of human speech recognition. Redundant information is removed from the signals thus slightly reducing the quality, but greatly reducing the required bandwidth.
In VoIP networks, the echo problem is typically compounded by the increased delay incurred by packetization of voice signals. To counteract the echo problem, VoIP gear (hard phones, soft phones, gateways) include echo cancelers to remove echo signals from the transmit signal.
To transport facsimile over an IP based network, even more technology is needed. Facsimile protocols are very susceptible to delay and delay variation and thus need more compensation algorithms. Protocols like T.38 terminate facsimile protocols like T.30 (analog facsimile) and transport the fax images as digitized pictures over IP based networks.
The document discusses GPON (Gigabit Passive Optical Network) technology and implementation models. It provides information on:
- GPON standards and components like the OLT, ONU, and splitters
- Implementation models for retail/residential, enterprise/HRB, and mobile backhaul networks
- Considerations for ODN design and link budget calculations for different splitting scenarios
- Capabilities of OLTs, ONUs, and ONTs including interfaces, services supported, and functionalities
- Examples of residential ODN installation and network architectures for different use cases
The document provides an overview of the public switched telephone network (PSTN). It discusses that the PSTN is the interconnected telephone system that uses copper wires to make circuit-switched calls. It then covers the evolution of the PSTN from its invention in 1876 to present digital switches, the use of bandwidth allocation and numbering schemes, and call setup which involves signaling and switching systems to route calls.
Wireless communication theodore rappaportDaud Khan
The document repeatedly lists the website "www.vsofts.net" and the word "oldroad" without any other context or information provided. It is not possible to determine the essential meaning or purpose of the document from the limited information given.
This document defines and compares two types of wireless local loop (WLL) technologies: Local Multipoint Distribution Service (LMDS) and Multichannel Multipoint Distribution Service (MMDS). LMDS operates above 20 GHz and provides high-speed broadband, while MMDS operates between 2.1-2.7 GHz and provides lower bandwidth but stronger signals over longer distances. Both can provide voice, data and video services as alternatives to wired local loops. Key differences are that LMDS supports higher data rates over shorter ranges while MMDS has a larger cell size and is more suitable for large networks.
Signaling System #7 (SS7) is a telecommunications protocol that defines high-speed circuit switching for telephone calls and uses out-of-band signaling between service switching points, signal transfer points, and service control points. It has advantages like separation of control information onto logically separate paths, message-oriented call information exchange, and ability of a single signaling channel to carry information about multiple trunks. The SS7 architecture includes service switching points, service control points, and signal transfer points that communicate using protocols like ISUP, TCAP, and SCCP.
- GPRS is an upgrade to GSM that allows packet-based data services and efficient use of network bandwidth. It provides higher data rates than GSM and constant connectivity.
- The GPRS network architecture introduces new network elements like the SGSN and GGSN to route data packets. The SGSN manages packet data in its service area while the GGSN connects the GPRS network to external packet networks.
- Session management in GPRS includes establishing PDP contexts for data transfer sessions and location management tracks the routing area of mobile devices through routing area updates.
This document provides an overview of Bluetooth technology, including its history, specifications, networks, layers, applications, and issues. Bluetooth was developed in the late 1990s to facilitate short-range wireless connectivity between devices. It uses radio waves and frequency hopping to transmit data between devices within a personal area network. Common applications of Bluetooth technology include connecting headphones, printers, and automobiles. While scalability and throughput are limitations, Bluetooth provides a simple, inexpensive way to connect electronic devices without wires.
Contents:
Data Traffic
Congestion
Congestion Control
Quality of Service
Techniques to improve QOS
How QOS is implemented within the Internet
References..
The document provides an overview of LTE (Long Term Evolution) network architecture and transmission schemes. It describes the simplified LTE network elements including eNB, MME, S-GW and P-GW. It explains the downlink transmission scheme using OFDMA and reference signal structure. It also covers uplink transmission using SC-FDMA, control and data channels as well as frame structure in both FDD and TDD modes.
This document summarizes information about video conferencing using H.323 standards. It discusses the H.323 protocols for audio and video, requirements for networks supporting H.323, and tools for diagnosing problems. It also provides an overview of ViDe.Net, an international virtual network providing video conferencing, and future directions for middleware to support directories and management of video users.
VoIP allows users to make phone calls using an Internet connection instead of a traditional phone line. It works by converting voice signals to digital data that is transmitted in packets over the Internet. A VoIP network uses protocols like SIP and RTP to setup calls and transmit voice data. Components include VoIP protocols, gateways to interface with the PSTN, and codecs to compress voice signals. Businesses are attracted to VoIP as it can help reduce costs while improving utilization of bandwidth and network management. However, security risks like hacking and eavesdropping exist since VoIP uses the public Internet.
SIP is an application-layer protocol for establishing multimedia sessions over IP networks. It can be used to initiate voice, video, and instant messaging communications. SIP works by having user agents (clients and servers) exchange SIP request and response messages. These messages contain information about session setup, modification, and termination. Some key SIP components include user agents, proxy servers, registrar servers, and redirect servers. SIP messages use a request-response transaction model and contain start lines, headers, and optional message bodies. Common request methods are INVITE, ACK, BYE, and REGISTER. Typical response codes include 100-199 (informational), 200-299 (success), 300-399 (redirection), 400-499
Introduction to SIP(Session Initiation Protocol)William Lee
Session Initiation Protocol (SIP) is a signaling protocol for managing multimedia communication sessions over Internet Protocol (IP) networks. SIP can be used to establish two-party or multiparty sessions that include voice, video, chat, gaming, and other forms of media. The document introduces SIP architecture, message format, and common call flows including registration, basic call setup, call modification, call hold, and three-way conferencing.
VoIP stands for Voice over Internet Protocol. It allows users to make phone calls using an IP network rather than a traditional telephone network. VoIP works by converting voice into packets of data that travel over the internet through routers to reach the destination. While it is beginning to be used more in businesses due to lower costs, some reliability issues with lost data packets can cause jittering and lower sound quality compared to traditional phone networks.
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.
SIP is a protocol for establishing multimedia sessions over IP networks. It originated from work in the 1990s on protocols like SCIP and SIP drafts. SIP eventually became standardized as RFC 3261 and is now widely used for voice and video calling. Cisco supports SIP in products like Cisco Unified Communications Manager, Cisco Unified Border Element, and Cisco Unified Presence to enable VoIP calling and integration between SIP and other protocols. The future of SIP includes more peer-to-peer implementations and using presence as a foundation for new services.
Voice over Internet Protocol (VoIP) allows users to make voice calls using an internet connection rather than a regular phone line. It works by encoding voice input and transmitting it as data packets over the internet. VoIP provides several benefits including lower costs, portability through mobile apps, and additional features like video calling. However, it also has some disadvantages like potential quality issues when making international calls and reliance on an internet connection to place calls.
The document discusses strategies for WiMAX radio network planning, focusing on balancing coverage and capacity. It covers designing coverage planning rules considering link budgets, frequency plans, cell radius calculations and prediction plots. Capacity planning strategies include setting subscriber and data rate projections over time and determining required base station configurations and densities. In-building solutions such as picocells and femtocells are also discussed as an approach to augment capacity.
This document provides an overview of Voice over Internet Protocol (VoIP) technology. It describes how VoIP works by converting voice signals to digital data that is transmitted over the Internet using packet switching. Common VoIP protocols like SIP and H.323 are discussed along with VoIP components like softphones, gateways, and codecs. Advantages of VoIP include low cost and flexibility, while disadvantages include reliability issues and lack of service during power outages. The document recommends that most VoIP issues will be addressed by 2008 when it will gain widespread consumer acceptance.
O documento descreve os principais serviços de streaming no Brasil, incluindo VOD (Video on Demand), TV Everywhere e serviços como Now, Netflix, Muu, Telecine Play, HBO GO, Crackle e Ever. Estes serviços oferecem filmes, séries e outros conteúdos para assistir online em diferentes dispositivos.
Poster Presentation of the 3rd IEEE Int. Conf. on ICIEV’14Habibur Rahman
The vehicular safety message feature is applied to avoid accident or collision avoidance on each vehicle. Analyzed the impact of IDM-IM and IDM-LC on AODV, AOMDV, DSDV and OLSR routing protocols in an urban scenario of Dhaka city. Recommend several concerns (drop rate, delay, jitter, route cost) before developing a realistic vehicular safety applications.
VoIPER: Smashing the VoIP stack while you sleepguestad6e9e
Voiper is an open source, cross-platform VoIP fuzzing toolkit that automates the process of finding vulnerabilities in SIP and SDP protocol implementations. It includes around 10 ready-to-use fuzzers covering the RFCs, a SIP library to manage sessions, and tools for crash detection and recreation to facilitate remote code execution testing without constant user interaction. Initial testing with Voiper found several crashes across four VoIP clients, demonstrating its ability to effectively hunt for bugs in SIP devices.
Presentation cloud telephony & ivrs based daily monitoring system from sudhan...stripathi_99
The document describes a cloud telephony and IVR-based daily monitoring system implemented in Uttar Pradesh, India to track the number of students receiving meals through the Mid Day Meal scheme across 1.52 lakh schools. Teachers provide data on the number of students fed daily by responding to automated IVR calls on their mobile phones. This provides real-time data and addresses issues like delays and data manipulation in the previous monthly reporting system. The new system has led to timely intervention, increased transparency, and a reduction in the number of schools not serving meals.
The Evolution of the Contact Center: Part 3 of 5Five9
The document summarizes the evolution of contact centers from the 1980s through mid-1990s, including:
1) The introduction of the internet in the 1980s brought a new channel for customer interactions through one-on-one online chats.
2) In the 1990s, contact centers began adopting multichannel environments enabled by widespread internet adoption, leading to the terminology changing from "call" centers to "contact" centers.
3) CRM software was also developed in the 1990s to help businesses create better customer experiences and implement rewards programs for repeat customers.
1. The document discusses authentication and user authentication. It covers various authentication mechanisms like passwords, biometrics, and one-time passwords.
2. Passwords are a common authentication method where the user supplies a password that is validated by the computer. Issues like password selection, storage, and cracking techniques are described.
3. Other authentication topics covered include encrypted password files, password guessing approaches, and ways to strengthen authentication like challenge-response systems and using biometrics instead of passwords.
This document outlines the topics that will be covered in a course on security operations centers (SOCs) and security information and event management (SIEM). It discusses traditional security approaches and their weaknesses. It then introduces advanced persistent threats, targeted malware like Stuxnet and Flame, and new mobile threats. The role of SIEM technologies and SOC frameworks for centralized security monitoring, analytics, and response are explained. Key components of SOCs like threat management, vulnerability management, and security intelligence services are also outlined.
Understand what are the causes and effects of the increased use of “Video on Demand”:
- Internet Access and Audience
- USA Crisis and Cable TV Fee
- Netflix VOD Operation and Content Negotiation
- Cable TV own VOD products and Market competition
This document discusses internet telephony and voice over internet protocol (VoIP). It describes the minimum requirements to use internet telephony including a PC, sound card, speakers, and internet connection. It explains how VoIP works by converting analog voice signals to digital data. There are three main types of internet telephony: analog telephone adaptor, IP phones, and PC-to-PC. Internet telephony offers lower costs for domestic and international calls compared to traditional phone services. While it provides benefits like reduced costs, security and reliability issues still exist.
H.323 is a multimedia conferencing protocol that allows for voice, video, and data conferencing over packet-switched networks like the internet. It defines standards for real-time communications including audio and video codecs, call signaling, and bandwidth management. H.323 has progressed through several versions to improve security, performance, and integration with other networks. Key components of an H.323 system include terminals for users, multipoint control units to enable conferencing, gateways to connect to other networks, and gatekeepers for call routing and access control. H.323 provides a foundation for unified communication services over IP networks.
The document discusses H.323 and related VoIP protocols. It provides an overview of H.323, including its history and evolution through various versions. It also discusses related protocols like SIP and how H.323 interconnects with other networks. Finally, it outlines security considerations and the future of multimedia communications.
H.323 is a standard that defines real-time multimedia communications over packet-based networks. It includes protocols for call signaling (H.225.0), media transport and control (RTP/RTCP, H.245), and bandwidth management. Key components include terminals, gateways, multipoint control units, and gatekeepers. Terminals support audio and optionally video/data. Gateways connect different networks. Multipoint control units enable video conferences. Gatekeepers provide address translation, admission control, and routing capabilities.
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.
Chuyên cung cấp giải pháp VOIP, 1800, 1900 cho doanh nghiệp
TIME TRUE LIFE TECHNOLOGY JOINT STOCK COMPANY
Mr Long
Mobi: 0986883886 - 0905710588
Email: long.npb@ttlcorp.vn
Website: ttlcorp.vn
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.
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.
Performance Analysis between H.323 and SIP over VoIPijtsrd
There are a number of protocols that may be employed in order to provide the Voice over IP VoIP communication services. In VoIP system, H.323 and Session Initiation Protocol SIP are the two major standards. Both of these signaling protocols provide mechanisms for multimedia teleconferencing services. Although the two protocols architecture is quite similar, they have many differences. This system presents Voice Video over IP communication and summarizes the differences and performance of two major VoIP protocols, H.323 and SIP according to the packet delay variation, jitter, packet loss, and Packet end to end delay. It is found that both of them are non interoperable, approaching each other, their focus and applicability is still different. In this paper, the system is designed and configured by Graphical Network Simulator GNS3 and analyzed performance by Opnet Modeler Simulation. Thet Zaw Aye "Performance Analysis between H.323 and SIP over VoIP" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-3 | Issue-5 , August 2019, URL: https://www.ijtsrd.com/papers/ijtsrd26647.pdfPaper URL: https://www.ijtsrd.com/engineering/computer-engineering/26647/performance-analysis-between-h323-and-sip-over-voip/thet-zaw-aye
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.
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 defines VoIP and its protocols. VoIP allows routing of phone calls over IP networks using packet switching instead of traditional circuit switching. Key protocols include H.323 for call setup, control and transport, and RTP for real-time media transport over UDP. H.323 defines codecs for digitizing and compressing voice, and uses signaling protocols like H.225 and H.245 as well as gatekeepers for call routing and quality of service control. RTCP monitors RTP transport quality. H.323 gateways enable interworking between IP and circuit-switched networks.
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.
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) 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.
- 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
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.
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 summarizes the design and implementation of a conference gateway that supports interoperability between the H.323 and SIP signaling protocols for multiparty video conferencing. Key points:
- The gateway allows H.323 and SIP clients to participate in the same conferencing session over an IP network. It acts as a gatekeeper, multipoint controller, and proxy server to translate between the protocols.
- Design decisions addressed mapping conference IDs, capability descriptions, membership control, and ongoing conference information between H.323 and SIP. Media streams are forwarded via the gateway without transcoding.
- An implementation was tested with up to 5 clients on each side. Performance evaluation showed comparable call setup times between the protocols ro
This document provides an overview of IP telephony and the H.323 standard. It describes the different components of H.323 including terminals, gateways, gatekeepers, and MCUs. It explains the protocols specified by H.323 such as H.225, H.245, RTP, and RTCP. It also provides high-level descriptions of H.323 call establishment, control signaling flows, media streams, and call release. The document concludes that IP telephony provides cost savings and new services by combining telephone and data networks into a powerful and economical communication option.
This document summarizes the H.323 standard for transmitting real-time audio, video, and data over IP networks. It describes the key components of H.323 including terminals, gateways, and gatekeepers. It also outlines the protocols used in H.323 like H.225 for call signaling and H.245 for capabilities exchange. The document explains the H.323 architectural model and provides a high-level overview of how H.323 calls are established and terminated between endpoints.
A version of watershed algorithm for color image segmentationHabibur Rahman
The document summarizes a master's thesis presentation on a new watershed algorithm for color image segmentation. The thesis addresses issues with existing watershed algorithms like over-segmentation and sensitivity to noise. The contributions of the thesis include an adaptive masking and thresholding mechanism to overcome over-segmentation and perform well on noisy images. The thesis is evaluated using five image quality assessment metrics on 20 classes of images, showing the proposed method performs better and has lower computational complexity than other algorithms. In conclusions, the adaptive watershed algorithm ensures accurate segmentation and is suitable for real-time applications.
Segmentation of Color Image using Adaptive Thresholding and Masking with Wate...Habibur Rahman
The document proposes a modified watershed algorithm for image segmentation. It applies adaptive masking and thresholding to each color channel before combining the results. The modified algorithm is compared to FCM, RG, and HKM using metrics like PSNR, MSE, PSNRRGB, and CQM on 10 images. Results show the proposed method ensures accuracy and quality while being faster than other algorithms, making it suitable for real-time use. It performs better than the other algorithms according to visual and quantitative analysis.
GreenCloud is an open source cloud computing simulator that allows researchers to observe and measure cloud performance. It is an extension of the NS2 network simulator and focuses on simulating communications within a cloud at the packet level. The tutorial discusses installing GreenCloud in Ubuntu either through a preconfigured virtual machine, or by manually downloading and compiling the source code. It also provides an example of running a sample simulation script.
This document provides an overview and tutorial on using CloudSim, an open-source simulation toolkit for modeling and simulation of cloud computing infrastructures and applications. It discusses CloudSim's features and architecture, prerequisites for using it, and how to set up the development environment in Eclipse. Sample code examples are presented to demonstrate running simulations of data centers with hosts and cloudlets using CloudSim.
This document provides an overview of several cloud simulation tools: CloudSim, CloudAnalyst, GreenCloud, and iCanCloud. CloudSim enables modeling and simulation of cloud computing infrastructures and applications. CloudAnalyst focuses on simulating large-scale cloud applications and studying their behavior under different deployment configurations using a graphical user interface. GreenCloud extends the NS2 network simulator to enable energy-aware cloud computing simulations at the packet level. iCanCloud allows modeling both existing and non-existing cloud architectures through a flexible hypervisor module and graphical interface to simulate distributed systems.
This document presents an example of discrete event simulation of customer arrivals and service at a bar called Bagha. It describes the simulation process, including modeling service times and inter-arrival times as random variables, maintaining an event list to track future events, and calculating various performance measures like average wait time at the end of the simulation. It also discusses how simulation can be used to study systems, evaluate design changes without disrupting real operations, and answer questions about performance.
Mobile IPv4 and Mobile IPv6 were simulated in NS2 to compare their performance. Packet drop and throughput were used as metrics. For TCP traffic over 200 seconds: MIPv4 had 3 dropped packets while MIPv6 had none. MIPv4 throughput was 13.12Mbps receiving, while MIPv6 was unlimited at 2.14Gbps receiving. In conclusion, MIPv6 performed better than MIPv4 according to the simulation results.
Localization with mobile anchor points in wireless sensor networksHabibur Rahman
The document describes a range-free localization scheme that uses mobile anchor points equipped with GPS to periodically broadcast their positions. Sensor nodes calculate their own positions based on the localization information from at least three mobile anchors without needing additional interactions. Simulation results showed the approach achieved fine-grained accuracy and was distributed, scalable, effective and power efficient. It performed better than other range-free localization mechanisms.
Directed diffusion for wireless sensor networkingHabibur Rahman
This document summarizes the key ideas of the "Directed Diffusion for Wireless Sensor Networking" paper. It introduces directed diffusion as a data-centric paradigm for wireless sensor networks that is designed for robustness, scalability, and energy efficiency. The core concepts of directed diffusion are interests, data, gradients, and reinforcement, which work together to efficiently route queries to sensor data in the network. Through localized interactions and data aggregation, directed diffusion is shown to significantly reduce energy consumption compared to flooding-based approaches in wireless sensor networks.
This presentation was provided by Rebecca Benner, Ph.D., of the American Society of Anesthesiologists, for the second session of NISO's 2024 Training Series "DEIA in the Scholarly Landscape." Session Two: 'Expanding Pathways to Publishing Careers,' was held June 13, 2024.
How Barcodes Can Be Leveraged Within Odoo 17Celine George
In this presentation, we will explore how barcodes can be leveraged within Odoo 17 to streamline our manufacturing processes. We will cover the configuration steps, how to utilize barcodes in different manufacturing scenarios, and the overall benefits of implementing this technology.
CapTechTalks Webinar Slides June 2024 Donovan Wright.pptxCapitolTechU
Slides from a Capitol Technology University webinar held June 20, 2024. The webinar featured Dr. Donovan Wright, presenting on the Department of Defense Digital Transformation.
Leveraging Generative AI to Drive Nonprofit InnovationTechSoup
In this webinar, participants learned how to utilize Generative AI to streamline operations and elevate member engagement. Amazon Web Service experts provided a customer specific use cases and dived into low/no-code tools that are quick and easy to deploy through Amazon Web Service (AWS.)
This document provides an overview of wound healing, its functions, stages, mechanisms, factors affecting it, and complications.
A wound is a break in the integrity of the skin or tissues, which may be associated with disruption of the structure and function.
Healing is the body’s response to injury in an attempt to restore normal structure and functions.
Healing can occur in two ways: Regeneration and Repair
There are 4 phases of wound healing: hemostasis, inflammation, proliferation, and remodeling. This document also describes the mechanism of wound healing. Factors that affect healing include infection, uncontrolled diabetes, poor nutrition, age, anemia, the presence of foreign bodies, etc.
Complications of wound healing like infection, hyperpigmentation of scar, contractures, and keloid formation.
How to Download & Install Module From the Odoo App Store in Odoo 17Celine George
Custom modules offer the flexibility to extend Odoo's capabilities, address unique requirements, and optimize workflows to align seamlessly with your organization's processes. By leveraging custom modules, businesses can unlock greater efficiency, productivity, and innovation, empowering them to stay competitive in today's dynamic market landscape. In this tutorial, we'll guide you step by step on how to easily download and install modules from the Odoo App Store.
How to Manage Reception Report in Odoo 17Celine George
A business may deal with both sales and purchases occasionally. They buy things from vendors and then sell them to their customers. Such dealings can be confusing at times. Because multiple clients may inquire about the same product at the same time, after purchasing those products, customers must be assigned to them. Odoo has a tool called Reception Report that can be used to complete this assignment. By enabling this, a reception report comes automatically after confirming a receipt, from which we can assign products to orders.
Andreas Schleicher presents PISA 2022 Volume III - Creative Thinking - 18 Jun...EduSkills OECD
Andreas Schleicher, Director of Education and Skills at the OECD presents at the launch of PISA 2022 Volume III - Creative Minds, Creative Schools on 18 June 2024.
Gender and Mental Health - Counselling and Family Therapy Applications and In...PsychoTech Services
A proprietary approach developed by bringing together the best of learning theories from Psychology, design principles from the world of visualization, and pedagogical methods from over a decade of training experience, that enables you to: Learn better, faster!
2. OutlineOutline
• What is H.323What is H.323
• Scope of H.323Scope of H.323
• Why is H.323 ImportantWhy is H.323 Important
• Historical Development StagesHistorical Development Stages
• Elements of H.323 SystemElements of H.323 System
• H.323 Network ArchitectureH.323 Network Architecture
• H.323 Core ProtocolsH.323 Core Protocols
• H.323: Call SignalingH.323: Call Signaling
• Prospect/Future of H.323Prospect/Future of H.323
2
3. What is H.323What is H.323
H.323* is a multimedia conferencing protocol, which
includes voice, video and data conferencing for use
over packet-switched networks
Real-time multimedia communications and
conferencing for packet-based networks
* H.323 is “ITU-T Recommendation H.323: Packet-based multimedia
communications systems”
3
4. Scope of H.323Scope of H.323
• Point-to-point and multipoint conferencing
support
• Inter-network interoperability
• Heterogeneous client capabilities
• Audio and video codecs
• Management and accounting support
• Security
• Supplementary services
4
5. Scope of H.323Scope of H.323
5
T1524040-96
Video I/O equipment
Audio I/O equipment
User Data Applications
T.120, etc.
System Control
User Interface
Video Codec
H.261, H.263
Audio Codec
G.711, G.722,
G.723, G.728,
G.729
System Control
H.245 Control
Call Control
H.225.0
RAS Control
H.225.0
Receive
Path
Delay
H.225.0
Layer
Network
Interface
Scope of Rec. H.323
6. Why is H.323 ImportantWhy is H.323 Important
TrendTrend
Rapid growth of the Internet
Universal deployment of corporate LANs have made
packet-based networks ubiquitous
StandardizationStandardization
H.323 is a standard protocol has been widely accepted
Promotes greater awareness, availability, and
acceptability of multimedia conferencing over packet-
based networks
6
7. Why is H.323 ImportantWhy is H.323 Important
Internet workingInternet working
Bridges multimedia communications between packet-
based and switched-circuit networks (SCN)
SCN conferencing standards like H.320 (ISDN), H.321
(ATM), and H.324 (PSTN) can inter-operate with H.323
clients
Integrated servicesIntegrated services
Additional services such as e-mail, voice mail, fax, call
center functionality and video conferencing in an
integrated environment
7
8. Why is H.323 ImportantWhy is H.323 Important
InteroperabilityInteroperability
ConnectivityConnectivity
ETSI/ IMTC
ITU-T/ IETF
Recommendations
ProductsProductsCustomers
V
E
N
D
O
R
S
StandardsStandards
8
9. Development Stages of H.323Development Stages of H.323
• H.323v1 published in 1996 & designed for LAN
– Companies tried to do use H.323 in WAN, large
private VoIP networks, and the Internet
• Guess what?
• It worked very well
• H.323 was an early adopter of IETF protocols as
RTP proved ability to carry real-time audio and
video over IP networks
– Indeed, H.323 was much more than a LAN
protocol name was changed in H.323 V2 (1998)
9
10. Development Stages of H.323 (con)Development Stages of H.323 (con)
Recognizing the fact that H.323 was much more
than a LAN protocol, the name was changed in
H.323 Version 2 (1998)
Enhancements were made, including:
Security
Performance
Supplementary Services
Scalability
10
11. Development Stages of H.323 (con)Development Stages of H.323 (con)
H.323 v3 introduced a few modest improvements,
mostly geared for better PSTN integration and
scalability
New annexes were introduced:
H.323 – UDP signaling
H.323 – Simple endpoint type
H.225.0 – Communication between
administrative domains
11
12. Development Stages of H.323 (con)Development Stages of H.323 (con)
• H.323 v4 was approved Nov. 2000 and brings a
number of enhancements to H.323. Areas of focus
include:
– Scalability
– Services
– “Must Have” Features
– Generic Extensibility Framework
• Current version of H.323 commonly referred to as
"H.323v6” was published in 2006
12
14. Scalability (con)Scalability (con)
Endpoint Capacity Reporting
By utilize endpoint capacity reporting, Gatekeepers may
select an endpoint that is best capable of handling the
call
This is extremely useful for large scale deployments of
Gateways and is also useful in call center applications
Never Lose a Call!
GK GK GK GK GK
GW
23%
GW
77%
GW
48%
GW
64%
GW
14%
GW
36%
The GK selects the GW with the most
capacity. H.323 endpoints report capacity
in absolute terms, not in percentages.
14
15. ServicesServices
Annex K – Services via HTTP
Annex L – Stimulus Control
H.450.8 – Name identification
H.450.9 – Call Completion
15
17. Generic Extensibility FrameworkGeneric Extensibility Framework
The Generic Extensibility Framework (GEF)
introduces a new means by which H.323 may be
further enhanced or extended with optional
features, which does not require changes to the
current ASN.1 syntax
Work has already begun
Robustness procedures (Annex R)
Local number portability
17
18. Components of H.323 SystemComponents of H.323 System
Terminals
what people see/hear
Multipoint Control Units (MCUs)
provides conference capabilities
Gateways
control and ‘routing’
Gatekeeper
access to other environments
Border Elements
18
20. TerminalsTerminals
An endpoint on the network which provides for
real-time, two-way communication with other
H.323 terminal, GW, or MCU
Terminal can be:
Telephones
Video phones
IVR devices
Voicemail Systems
“Soft phones” (e.g., NetMeeting®)
20
T
21. Multipoint Control Units (MCUs)Multipoint Control Units (MCUs)
Needed
only when multiparty
conferences are desired
Functions:
To manages call
signaling
Provides capability of
videoconferencing with
more than one party
Acts as a coordinator of
multiparty conferences
2121
22. Gateways (GW)Gateways (GW)
• Gateway (GW)
• used as interface H.323
between different networks
e.g. LAN & PSTN
• Functions:
• Data format translation
• Audio/video codec
translation
• Call setup, termination
from both sides of the
network
22
23. Gatekeeper (GK)Gatekeeper (GK)
Gatekeeper is an optional component in H.323 system
used for:
Admission Control and
Address Resolution
Endpoints do register themselves at a Gatekeeper
All H.323 endpoints registered to a single GK build an
H.323 zone
H.323 zones are independent of physical network
topology
Each zone has only one GK (exception: Alternate
GKs)
23
26. Border Elements (BE)Border Elements (BE)
Co-located with Gatekeeper, Exchange addressing
information
Participate in call authorization between
administrative domains
May aggregate address information to reduce the
volume of routing information passed through the
network
May assist in call authorization/authentication
directly between two administrative domains or
via a clearinghouse
26
27. Using Elements (BE)Using Elements (BE)
27
As with hierarchical
Gatekeepers, Border
Elements may send Access
Request messages to other
Border Elements and
indicate where to send a
reply
Border Elements may also
reply directly to a request by
utilizing address
information cached from
previous exchanges with
other Border Elements
T
GK
LRQ
GK/BE
ARQ
GK/BE
AccessRequest
29. Protocols of H.323Protocols of H.323
H.323 is an umbrella of four protocols:
• Registration Admission and Status (RAS)
– define communications between endpoints and gatekeeper
– only needed when a gatekeeper exists
• H.245 - Connection Control for Capability Negotiations
• H.225/Q.931- Call Signaling (between endpoint and gatekeeper, or
between gatekeepers)
• Real-time Transport Protocol(RTP) - timely and orderly
delivery of audio and video streams
29
31. Registration Admission and Status (RAS)Registration Admission and Status (RAS)
• Defined in H.225.0
• Allows an endpoint to request authorization to
place or accept a call
• Allows a Gatekeeper to control access to and from
devices under its control
• Allows a Gatekeeper to communicate the address
of other endpoints
• Allows two Gatekeepers to easily exchange
addressing
31
32. Registration Admission and Status (con)Registration Admission and Status (con)
32
T GKRRQ
RCF
ARQ
(endpoint is registered)
ACF
(endpoint may place call)
DRQ
DCF
(call has terminated)
33. H.225H.225
H.225 Call Signaling
H.225 call signaling is used to establish a connection
between two H.323 endpoints
Achieved by exchanging H.225 protocol messages on
the call-signaling channel
call-signaling channel is opened between two H.323
endpoints or between an endpoint and gatekeeper
H.225 is the conference control protocol
Master/slave determination
Capability exchange
Management of media and data streams
33
34. RTP/RTCPRTP/RTCP
RTP/RTCP used for audio & video over IP
networks
H.225 call signaling is used to establish a connection
between two H.323 endpoints
Achieved by exchanging H.225 protocol messages on
the call-signaling channel
call-signaling channel is opened between two H.323
endpoints or between an endpoint and gatekeeper
Real Time Transport Protocol (RTP)
end-to-end network transport function
payload type, sequence number, timestamp
RTP Control Protocol (RTCP)
34
35. CODECsCODECs
Audio
G.711 (popular codec for telephone n/ws)
G.723.1 – more efficient
Video
H.261 codec (for channels with bandwidths p*64
kb/s)
H.263 codec (for low bit rate transmission without
loss of quality )
35
36. Voice over IPVoice over IP
36
Voice over IP (VoIP or Voice over Internet Protocol)
commonly refers to the communication protocols,
technologies, methodologies, and transmission techniques
involved in the delivery of voice communications and
multimedia sessions over Internet Protocol (IP) networks,
such as the Internet.
Other terms commonly associated with VoIP are:
IP telephony, Internet telephony, voice over broadband
(VoBB), broadband telephony, IP communications, and
broadband phone.
37. Voice over IP: ProtocolsVoice over IP: Protocols
37
Voice over IP has been implemented in various ways
using both proprietary and open protocols and
standards. Examples of the network protocols used to
implement VoIP include:
H.323
Media Gateway Control Protocol (MGCP)
Session Initiation Protocol (SIP)
Real-time Transport Protocol (RTP)
Session Description Protocol (SDP)
Inter-Asterisk eXchange (IAX)
38. Voice over IP: Protocols (con)Voice over IP: Protocols (con)
38
H.323 protocol was one of the first VoIP protocols
found widespread implementation for long-distance
traffic, as well as local area network services.
However, since the development of newer, less
complex protocols such as MGCP and SIP, H.323
deployments are increasingly limited to carrying
existing long-haul network traffic.
Session Initiation Protocol (SIP) has gained
widespread VoIP market penetration.
A notable proprietary implementation is the Skype
protocol, which is in part based on the principles of
peer-to-peer (P2P) networking.
39. Voice over IP: Business useVoice over IP: Business use
39
40. Session Initiation Protocol (SIP)Session Initiation Protocol (SIP)
40
Session Initiation Protocol (SIP) is an Internet
Engineering Task Force (IETF) standard protocol for
initiating an interactive user session that involves
multimedia elements such as video, voice, chat,
gaming, and virtual reality.
The protocol can be used for creating, modifying and
terminating two-party (unicast) or multiparty
(multicast) sessions. Sessions may consist of one or
several media streams.
Other SIP applications include video conferencing,
streaming multimedia distribution, instant
messaging, presence information, file transfer.
46. H.323 Protocol ArchitectureH.323 Protocol Architecture
ControlControl DataData Audio Video AV Control GK ControlAudio Video AV Control GK Control
signal + connection
46
Q.931Q.931
/H.22/H.22
55
H.245H.245 RTCPRTCPT.120T.120 RASRAS
H.26xH.26xG.7xxG.7xx
RTPRTP
UDPUDP
IPIP
TCPTCP
Protocol Relationships in H.323
51. Basic Call set-up with No GatekeeperBasic Call set-up with No Gatekeeper
51
t
52. Call set-up with Gatekeeper RoutingCall set-up with Gatekeeper Routing
t
52
Connect
ARQ
ACF
ACF
Alerting
Alerting
Set-upSet-up
Call Presiding
Connect
ARQ
GW GWGK
53. Call set-up with Gatekeeper RoutingCall set-up with Gatekeeper Routing
t
53
54. Security Issue in H.323Security Issue in H.323
in H.323 v1
H.235 Security protocols ITU
authentication: end-point authentication
integrity: validation within a packet
privacy: encryption and decryption mechanism
non-repudiation: false denial of participation
54
55. Call Enhancement in H.323Call Enhancement in H.323
H.323 v2 & v3
H.450 on top of Q.931
H.450.1 – Generic functional protocols and procedures [v1]
H.450.2 – Call Transfer [v2]
H.450.3 – Call Diversion
H.450.4 – Call Hold
H.450.5 – Call Park and Pick-up [v3]
H.450.6 – Message Waiting indication
H.450.7 – Call Waiting
H.450.8 – Name Identification[v4]
H.450.9 – Call Completion
H.450.10 – Call Offer
H.450.11 – Call Intrusion and so on…
55
56. H.323: Market TodayH.323: Market Today
Today the biggest market for H.323 applications is
Voice over IP.
Why?
Low bit-rate Internet connections make video and data
intensive applications less appealing
It’s a young industry– and with all such industries, it
takes time to mature good products
Companies can provide VoIP services today at a low
cost and provide new competition to the incumbent
carriers
56
57. H.323: The Changing MarketH.323: The Changing Market
Tomorrow, expect to see video and data
conferencing to become more pervasive
Broadband connectivity is making it possible
Video and data are logically the next services
customers expect to find in conference rooms and on
their computer screens
57
58. H.323: Beyond Voice over IPH.323: Beyond Voice over IP
Voice over IP opens the door to the next generation
of communication products
It will take some time to migrate the world from
PSTN to IP networks
H.323 provides excellent interworking between IP
networks and the PSTN
H.323 provides a strong foundation for new multimedia
products and services
58
59. H.323: IP TelephonyH.323: IP Telephony
• IP Telephony with H.323 truly means Multimedia
over IP
• IP Telephony is not Just Research Topic Anymore
– is now real… there are many deployed products and
services that offer IP Telephony services
– new kinds of services are now available to customers
using IP Telephony that were never possible before
59
60. H.323: Makes All PossibleH.323: Makes All Possible
H.323 makes it possible to create and deploy new
services quickly and to take advantage of
multimedia capabilities
These services can embrace audio, video, and
data conferencing
- Application Sharing - Electronic Whiteboard - File Transfer
- Instant Messaging - Click to Dial - Internet Call Waiting
- Web Call Parking - Call No-Waiting - Ad-Hoc Conferencing
- Voicemail Anywhere - Unified Messaging - Service Portability
- Services! - Services! - Services!
60
61. H.323: for Service ProviderH.323: for Service Provider
H.323 is a proven technology that is utilized in
large networks, such as Genuity, iBasis, ITXC,
China Unicom, and others
Excellent integration with the PSTN
Gateways and residential devices are in use today
61
62. H.323: in the EnterpriseH.323: in the Enterprise
Multimedia conferencing devices show the real
potential of H.323 and multimedia communication
With H.323 in the service provider network, H.323 is
a logical choice for the enterprise
The enterprise customer wants voice, video, and data
conferencing capabilities
62
63. Importance of H.323Importance of H.323
Interoperability - H.323 establishes methods for receiving
clients to communicate capabilities to the sender
Network independence - H.323 is not tied to any
hardware or operating system
H.323 sets multimedia standards for the existing
infrastructure (i.e. IP-based networks)
H.323 conference can include endpoints with different
capabilities
H.323 provides multiple audio and video CODECs that
format data according to the requirements of various
networks, using different bit rates, delays, and quality
options.
63
64. Importance of H.323 (con)Importance of H.323 (con)
Although H.323 can support conferences of three or more
endpoints without requiring a specialized multipoint
control unit, MCU's provide a more powerful and flexible
architecture for hosting multipoint conferences
Although H.323 can support conferences of three or more
endpoints without requiring a specialized multipoint
control unit, MCU's provide a more powerful and flexible
architecture for hosting multipoint conferences
H.323 supports multicast transport in multipoint
conferences
H.323 has the support of many computing and
communications companies and organizations
64
66. Recommendation AnnexesRecommendation Annexes
H.323
Annex C – H.323 over ATM
Annex D – H.323 FAX
Annex E – UDP operation
Annex F – Simple Endpoints
Annex J – Security for Simple endpoints
Annex K – HTTP based call control
Annex M – Tunneling of QSIG in H.323
H.225.0
Annex G – Inter-Domain Communications
Annex H – ASN.1 Syntax
66
67. ConclusionConclusion
H.323 is a protocol that leverages the strength
of the packet-switched protocols from the
IETF
Offers excellent integration with the PSTN
H.323 enables voice, video, and data
conferencing
H.323 provides a solid foundation for new
services and the continued growth of
Multimedia over IP
67
68. AcronymsAcronyms
• ARQ –Admission Request message
• BE – Border Element
• GEF – Generic Extensibility Framework
• GK – Gatekeeper
• GW – Gateway
• IETF – Internet Engineering Task Force
• IMTC – International Multimedia Telecommunications
Consortium
• IP – Internet Protocol
• IVR – Interactive Voice Response
• LAN – Local Area Network
• LRQ – H.225.0 Location Request message
• MCU – Multipoint Control UnitMC – Multipoint Controller
• MG – Media Gateway
• MGC – Media Gateway Controller
• MP – Multipoint Processor
• PSTN – Public Switched Telephone Network
68
69. AcronymsAcronyms
• RFC – Request for Comments
• RTP – Real-Time Transport Protocol
• RTCP – Real-time Transport Control Protocol
• TCS – H.245 Terminal Capability Set message
• UDP – User Datagram Protocol
• URL – Uniform Resource Locator
• VoIP – Voice over IP
• MC – Multipoint Controller
• MG – Media Gateway
• MGC – Media Gateway Controller
• MP – Multipoint Processor
• PSTN – Public Switched Telephone Network
• RFC – Request for Comments
• RTP – Real-Time Transport Protocol
• RTCP – Real-time Transport Control Protocol
• TCS – H.245 Terminal Capability Set message
• UDP – User Datagram Protocol
• URL – Uniform Resource Locator
• VoIP – Voice over IP
69
ITXC - Internet Telephony Exchange Carrier (US based wholesale provider of VoIP China has adopted H.323 as their national standard for IP telephony communications