Ccnp collaboration plus module 1 chapter 8 dial plan and call routingFaisal Khan
This document provides an overview of dial plans and call routing in Cisco Collaboration networks. It describes key components of dial plans including dial peers, destination patterns, and digit manipulation. Dial peers establish logical connections between endpoints and define call properties. Destination patterns are used to match dialed numbers and select the appropriate outbound dial peer. The document also covers techniques for inbound and outbound dial peer matching and examples of digit consumption, forwarding, collection, and translation configurations.
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.
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.
VoIP is one of a family of internet technologies and transmission technologies for delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks.
FB: https://www.facebook.com/mstfahsin
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.
Pbx presentation ingate_itexpoeast2014kwader Saudi
Enhance employee productivity and reduce communication costs with feature-rich IP telephony solutions from Kwader. With our solutions, your staff can count on effective, unified communications no matter where they are.
KTC scalable IP telephony solutions offer the same high-quality communications whether your enterprise has a few or 100,000 users. Our flexible architecture design offers an unparalleled range of deployment options. Our wide range of resiliency tools minimizes costs and maximizes reliability.
SIP - More than meets the eye
Speakers:
Ofer Cohen - VOIP Group Leader, LivePerson
Yossi Maimon - VOIP Technical Leader, LivePerson
An Introduction to the SIP protocol.
SIP Position in telecommunication networks and the content services.
What is SIP:
The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating sessions consisting of one or several media streams. SIP can be used for two-party (unicast) or multiparty (multicast) sessions. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.
(Source: Wikipedia)
This document provides an overview of Voice over Internet Protocol (VoIP) technology. It discusses what VoIP is, how it works, its components, advantages like lower costs, and disadvantages like potential quality issues. The document also compares VoIP to the traditional Public Switched Telephone Network (PSTN) and explores VoIP alternatives and the future of the technology. Overall, the document serves as an introduction to VoIP and its capabilities for voice communication over the internet.
Ccnp collaboration plus module 1 chapter 8 dial plan and call routingFaisal Khan
This document provides an overview of dial plans and call routing in Cisco Collaboration networks. It describes key components of dial plans including dial peers, destination patterns, and digit manipulation. Dial peers establish logical connections between endpoints and define call properties. Destination patterns are used to match dialed numbers and select the appropriate outbound dial peer. The document also covers techniques for inbound and outbound dial peer matching and examples of digit consumption, forwarding, collection, and translation configurations.
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.
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.
VoIP is one of a family of internet technologies and transmission technologies for delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks.
FB: https://www.facebook.com/mstfahsin
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.
Pbx presentation ingate_itexpoeast2014kwader Saudi
Enhance employee productivity and reduce communication costs with feature-rich IP telephony solutions from Kwader. With our solutions, your staff can count on effective, unified communications no matter where they are.
KTC scalable IP telephony solutions offer the same high-quality communications whether your enterprise has a few or 100,000 users. Our flexible architecture design offers an unparalleled range of deployment options. Our wide range of resiliency tools minimizes costs and maximizes reliability.
SIP - More than meets the eye
Speakers:
Ofer Cohen - VOIP Group Leader, LivePerson
Yossi Maimon - VOIP Technical Leader, LivePerson
An Introduction to the SIP protocol.
SIP Position in telecommunication networks and the content services.
What is SIP:
The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating sessions consisting of one or several media streams. SIP can be used for two-party (unicast) or multiparty (multicast) sessions. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.
(Source: Wikipedia)
This document provides an overview of Voice over Internet Protocol (VoIP) technology. It discusses what VoIP is, how it works, its components, advantages like lower costs, and disadvantages like potential quality issues. The document also compares VoIP to the traditional Public Switched Telephone Network (PSTN) and explores VoIP alternatives and the future of the technology. Overall, the document serves as an introduction to VoIP and its capabilities for voice communication over the internet.
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.
SIP (Session Initiation Protocol) trunking connects a company's PBX to the existing telephone network infrastructure via the internet using VoIP. It was originally designed in 1996 and standardized in 2000. SIP trunking provides benefits like virtual phone numbers, reduced equipment needs, business continuity, and flexible trunk quantities. However, considerations must include system compatibility, additional bandwidth requirements, and challenges like supporting fax/modem traffic and 911 calls. The document discusses ideal environments for SIP trunking like companies with multiple locations, seasonal needs, or those seeking increased functionality. It also reviews cost components and provides an overview of SIP trunking.
The document is a presentation by BroadConnect Telecom that introduces VoIP technology and BroadConnect's VoIP-enabled products and services. It defines VoIP as the delivery of voice communications over IP networks using standards-based protocols. It then describes BroadConnect's SIP server, IP phones, gateways, audio/video conferencing units, and IP cameras. The presentation explains how VoIP works by using codecs like G.711 to transfer voice data over the IP network. It outlines VoIP's advantages of low cost and security, as well as its need for constant power and internet connectivity. BroadConnect provides hosted PBX, SIP trunking, phone lines, communication services, internet services, and hosting solutions to help businesses simplify communications
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 provides an overview of Voice over Internet Protocol (VoIP) through a seminar presentation covering what VoIP is, why and when to use it, how it works, its architecture and components, advantages, disadvantages, alternatives, and the future of VoIP. Key points include that VoIP allows routing of voice conversations over the internet or IP networks, it can provide cheaper telecommunications through reduced phone and wiring costs, and integrates features like video conferencing. Quality concerns and dependency on network hardware are disadvantages.
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.
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.
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.
SIP is a protocol for setting up and managing sessions over the internet, including voice and video calls. It allows users to locate each other and establish communication sessions between endpoints. SIP sets up sessions but does not handle the actual media, like audio, which is transported separately using protocols like RTP. SIP works by routing request and response messages between user agents through proxies and servers to initiate, negotiate, and terminate communication sessions.
Introduction to VoIP, 2nd chapter of "Unified Communications with Elastix" Vol.1
We recommend to read the chapter along with the presentation.
http://elx.ec/chapter2
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.
Over the past 10 years the Session Initiation Protocol (SIP) has moved from the toy of researchers and academics to the de-facto standard for telephony and multimedia services in mobile and fixed networks.
Probably one of the most emotionally fraught discussions in the context of SIP was whether Session Border Controllers (SBC) are good or evil.
SIP was designed with the vision of revolutionizing the way communication services are developed, deployed and operated. Following the end-to-end spirit of the Internet SIP was supposed to turn down the walled gardens of PSTN networks and free communication services from the grip of large telecom operators. By moving the intelligence to the end systems, developers were supposed to be able to develop new communication services that will innovate the way we communicate with each other.
This was to be achieved without having to wait for the approval of the various telecommunication standardization groups such as ETSI or the support of incumbent telecoms.
Session border controllers are usually implemented as SIP Back-to-Back User Agents (B2BUA) that are placed between a SIP user agent and a SIP proxy. The SBC then acts as the contact point for both the user agents and the proxy. Thereby the SBC actually breaks the end-to-end behavior of SIP, which has led various people to deem the SBC as an evil incarnation of the old telecom way of thinking. Regardless of this opposition, SBCs have become a central part of any SIP deployment.
In this paper we will first give a brief overview of how SIP works and continue with a description of what SBCs do and the different use cases for deploying SBCs.
The document outlines basic call flows for location updates, mobile originating calls (MOC), mobile terminating calls (MTC), and IP calls. It describes the key steps as:
1) Location update involves identity response, authentication between the SIM and MSC, update location requests, and ciphering.
2) For MOC, the mobile station sends a setup message with the dialed number, the MSC sends a send routing information message to the HLR, and the HLR responds with routing instructions allowing the call to be connected.
3) For MTC, the MSC requests a roaming number from the HLR, the HLR provides a number and the MSC pages the mobile station to alert
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.
The document discusses the Session Initiation Protocol (SIP), which allows for multimedia communication sessions over IP networks. SIP establishes sessions for voice, video, messaging and other applications. It uses requests and responses to initiate sessions between users, locate users, invite them to sessions, and terminate sessions. SIP relies on user agents, proxy servers, redirect servers and registrar servers. It enables mobility and flexibility in setting up and modifying communication sessions across different devices.
Session Initiation Protocol (SIP) is an application layer signaling protocol used to establish, modify, and terminate multimedia sessions over the Internet. SIP allows users to initiate and manage telephone calls, video conferences, messaging, and other multimedia sessions. It can be used with other protocols like SDP, RTP, and RTCP to build a complete multimedia architecture. SIP establishes sessions through proxy servers, redirect servers, and registrars, and uses response codes to indicate session status. A basic SIP call flow involves an INVITE request, provisional responses, a final 200 OK response, media transmission via RTP, and termination with a BYE request.
Cisco CallManager Express (CME) is a software-based call processing solution that provides voice over IP (VoIP) functionality for small-to-medium sized businesses. It allows for up to 120 IP phones and uses IOS on Cisco routers to provide call processing capabilities. CME utilizes various protocols like Skinny, H.323, and SIP to connect IP phones, integrate with the PSTN, and communicate between CME systems over a WAN. It also requires configuration of items like auxiliary VLANs and DHCP services to properly operate and assign IP addresses to phones.
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.
O Brasil é o maior produtor e exportador de café do mundo, e também destaca-se por fornecer cafés especiais de alta qualidade. O consumo desses cafés especiais vem crescendo rapidamente no país. A Itapema FM planeja realizar o projeto "Barista Itapema" em 2014 para promover o aprendizado e apreciação da cultura do café.
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.
SIP (Session Initiation Protocol) trunking connects a company's PBX to the existing telephone network infrastructure via the internet using VoIP. It was originally designed in 1996 and standardized in 2000. SIP trunking provides benefits like virtual phone numbers, reduced equipment needs, business continuity, and flexible trunk quantities. However, considerations must include system compatibility, additional bandwidth requirements, and challenges like supporting fax/modem traffic and 911 calls. The document discusses ideal environments for SIP trunking like companies with multiple locations, seasonal needs, or those seeking increased functionality. It also reviews cost components and provides an overview of SIP trunking.
The document is a presentation by BroadConnect Telecom that introduces VoIP technology and BroadConnect's VoIP-enabled products and services. It defines VoIP as the delivery of voice communications over IP networks using standards-based protocols. It then describes BroadConnect's SIP server, IP phones, gateways, audio/video conferencing units, and IP cameras. The presentation explains how VoIP works by using codecs like G.711 to transfer voice data over the IP network. It outlines VoIP's advantages of low cost and security, as well as its need for constant power and internet connectivity. BroadConnect provides hosted PBX, SIP trunking, phone lines, communication services, internet services, and hosting solutions to help businesses simplify communications
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 provides an overview of Voice over Internet Protocol (VoIP) through a seminar presentation covering what VoIP is, why and when to use it, how it works, its architecture and components, advantages, disadvantages, alternatives, and the future of VoIP. Key points include that VoIP allows routing of voice conversations over the internet or IP networks, it can provide cheaper telecommunications through reduced phone and wiring costs, and integrates features like video conferencing. Quality concerns and dependency on network hardware are disadvantages.
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.
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.
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.
SIP is a protocol for setting up and managing sessions over the internet, including voice and video calls. It allows users to locate each other and establish communication sessions between endpoints. SIP sets up sessions but does not handle the actual media, like audio, which is transported separately using protocols like RTP. SIP works by routing request and response messages between user agents through proxies and servers to initiate, negotiate, and terminate communication sessions.
Introduction to VoIP, 2nd chapter of "Unified Communications with Elastix" Vol.1
We recommend to read the chapter along with the presentation.
http://elx.ec/chapter2
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.
Over the past 10 years the Session Initiation Protocol (SIP) has moved from the toy of researchers and academics to the de-facto standard for telephony and multimedia services in mobile and fixed networks.
Probably one of the most emotionally fraught discussions in the context of SIP was whether Session Border Controllers (SBC) are good or evil.
SIP was designed with the vision of revolutionizing the way communication services are developed, deployed and operated. Following the end-to-end spirit of the Internet SIP was supposed to turn down the walled gardens of PSTN networks and free communication services from the grip of large telecom operators. By moving the intelligence to the end systems, developers were supposed to be able to develop new communication services that will innovate the way we communicate with each other.
This was to be achieved without having to wait for the approval of the various telecommunication standardization groups such as ETSI or the support of incumbent telecoms.
Session border controllers are usually implemented as SIP Back-to-Back User Agents (B2BUA) that are placed between a SIP user agent and a SIP proxy. The SBC then acts as the contact point for both the user agents and the proxy. Thereby the SBC actually breaks the end-to-end behavior of SIP, which has led various people to deem the SBC as an evil incarnation of the old telecom way of thinking. Regardless of this opposition, SBCs have become a central part of any SIP deployment.
In this paper we will first give a brief overview of how SIP works and continue with a description of what SBCs do and the different use cases for deploying SBCs.
The document outlines basic call flows for location updates, mobile originating calls (MOC), mobile terminating calls (MTC), and IP calls. It describes the key steps as:
1) Location update involves identity response, authentication between the SIM and MSC, update location requests, and ciphering.
2) For MOC, the mobile station sends a setup message with the dialed number, the MSC sends a send routing information message to the HLR, and the HLR responds with routing instructions allowing the call to be connected.
3) For MTC, the MSC requests a roaming number from the HLR, the HLR provides a number and the MSC pages the mobile station to alert
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.
The document discusses the Session Initiation Protocol (SIP), which allows for multimedia communication sessions over IP networks. SIP establishes sessions for voice, video, messaging and other applications. It uses requests and responses to initiate sessions between users, locate users, invite them to sessions, and terminate sessions. SIP relies on user agents, proxy servers, redirect servers and registrar servers. It enables mobility and flexibility in setting up and modifying communication sessions across different devices.
Session Initiation Protocol (SIP) is an application layer signaling protocol used to establish, modify, and terminate multimedia sessions over the Internet. SIP allows users to initiate and manage telephone calls, video conferences, messaging, and other multimedia sessions. It can be used with other protocols like SDP, RTP, and RTCP to build a complete multimedia architecture. SIP establishes sessions through proxy servers, redirect servers, and registrars, and uses response codes to indicate session status. A basic SIP call flow involves an INVITE request, provisional responses, a final 200 OK response, media transmission via RTP, and termination with a BYE request.
Cisco CallManager Express (CME) is a software-based call processing solution that provides voice over IP (VoIP) functionality for small-to-medium sized businesses. It allows for up to 120 IP phones and uses IOS on Cisco routers to provide call processing capabilities. CME utilizes various protocols like Skinny, H.323, and SIP to connect IP phones, integrate with the PSTN, and communicate between CME systems over a WAN. It also requires configuration of items like auxiliary VLANs and DHCP services to properly operate and assign IP addresses to phones.
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.
O Brasil é o maior produtor e exportador de café do mundo, e também destaca-se por fornecer cafés especiais de alta qualidade. O consumo desses cafés especiais vem crescendo rapidamente no país. A Itapema FM planeja realizar o projeto "Barista Itapema" em 2014 para promover o aprendizado e apreciação da cultura do café.
Dokumen tersebut membahas tentang komunikasi, meliputi definisi komunikasi, proses komunikasi, bentuk dan metode komunikasi, hambatan komunikasi, mendengarkan secara aktif, umpan balik, perilaku nonverbal, pedoman komunikasi yang efektif, dan model-model komunikasi.
O SBT Pará oferece vinhetas comemorativas de 25 segundos para datas especiais ao longo do ano, com 5 segundos de assinatura da empresa cliente para reforçar sua marca. O projeto pode ser adaptado para qualquer data comemorativa e relaciona a marca do cliente a grandes momentos.
O documento descreve as ações de divulgação e cobertura que as rádios Eldorado FM e Rádio Estadão farão do 12o Festival Literário de Paraty (FLIP) em 2014. O festival ocorrerá de 30 de julho a 3 de agosto e homenageará o escritor Millôr Fernandes. As rádios farão chamadas de divulgação, flashes ao vivo, entrevistas e boletins sobre o evento.
O documento descreve um projeto especial do portal iBahia para o Dia das Crianças de 2014, incluindo a criação de um canal dedicado com conteúdo para entretenimento infantil, promoções de presentes e programação cultural relacionada à data na Bahia. O projeto também inclui uma galeria de fotos das crianças dos leitores e detalhes sobre pacotes de patrocínio disponíveis para marcas.
O documento discute a importância da participação na gestão escolar. Afirma que a gestão escolar requer a participação conjunta de todos os membros da comunidade escolar, incluindo professores, funcionários e pais. Discutem-se diferentes níveis de participação e como práticas limitadas de participação podem ser ineficazes ou até prejudiciais. Valores como ética, solidariedade e compromisso são apontados como fundamentais para uma participação efetiva.
Proyecto sobre LEYENDAS ARGENTINAS, realizado por alumnos de la sala de 5 años, turno tarde, del Instituto Inmaculada Concepción de Nuestra Señora de Lourdes, Buenos Aires, Argentina.
Cada presentación corresponde a distintos grupos de trabajo de 5 o 6 alumnos, que fue materializada en el armado de los libros de las leyendas.
A rádio FM 93 é líder de audiência em Fortaleza há mais de 20 anos, com uma média de 3,92% de audiência e 122 mil ouvintes por minuto. Ela possui o maior alcance no estado, atingindo 1,2 milhão de ouvintes mensalmente, sendo 68% das classes ABC. A FM 93 fará a cobertura do Fortal, o maior carnaval fora de época de Fortaleza.
ISP ABC provides ADSL service to subscribers using IP addresses between 205.1.1.2-205.1.1.100. Subscribers use PPPoE authentication with PAP. The ISP uses DHCP to assign IP addresses to subscribers and NAT-overload to allow subscribers on the same LAN to access the internet.
This document provides instructions for configuring IPv6 addressing, routing, and basic connectivity on Cisco devices. It includes configuration steps for enabling IPv6 routing, defining IPv6 prefixes, configuring IPv6 on interfaces, and troubleshooting IPv6 connectivity issues. The document also contains examples of IPv6 addressing configurations.
1. The document discusses how Internet Protocol version 6 (IPv6) interacts with and improves upon IPv4.
2. It explains that IPv6 addresses some key limitations of IPv4 by providing more IP addresses and incorporating other improvements to routing and network architecture.
3. Specifically, it outlines how IPv6 can be used to transition networks currently using IPv4 in a dual-stack approach, allowing newer devices and networks to communicate using IPv6 while maintaining compatibility with IPv4.
1. The document discusses Internet Protocol version 6 (IPv6), which was developed by the Internet Engineering Task Force (IETF) to replace IPv4.
2. It explains the different types of IPv6 addresses: Unicast, Anycast, and Multicast. Unicast is for single interfaces, Anycast is for load sharing across multiple interfaces, and Multicast is for group communications.
3. It provides examples of IPv6 address formats and how the address fields are used for different types of local and site-local addresses.
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Biện pháp rèn kĩ năng diễn đạt trong văn miêu tả cho học sinh lớp 5.pdfTÀI LIỆU NGÀNH MAY
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CHẾ ĐỘ VUA LÊ - CHÚA TRỊNH, MỘT THỂ CHẾ CHÍNH TRỊ ĐẶC BIỆT TRONG LỊCH SỬ DÂN ...TÀI LIỆU NGÀNH MAY
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https://www.facebook.com/thuvienluanvan01
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tai lieu tong hop, thu vien luan van, luan van tong hop, do an chuyen nganh
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tai lieu tong hop, thu vien luan van, luan van tong hop, do an chuyen nganh
MỘT SỐ BIỆN PHÁP TẠO BIỂU TƯỢNG VĂN HÓA VẬT CHẤT TRONG DẠY HỌC LỊCH SỬ THẾ GI...TÀI LIỆU NGÀNH MAY
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1. Cấu hình gatekeeper với một Zone
Tóm tắt chức năng Gatekeeper
Gatekeeper là thiết bị đóng vai trò như thiết bị hỗ trợ thông tin định tuyến cuộc gọi trong
công nghệ VoIP, bản thân các gateway có thể chứa các thông tin giúp định tuyến các
cuộc gọi trực tiếp lẫn nhau, tuy nhiên khi số lượng VoIP gateway tăng dần thì việc cấu
hình các thông tin định tuyến này trở nên quá lớn và phân tán. Do đó, các thông tin định
tuyến cuộc gọi được cấu hình tập trung trên gatekeeper, khi các gateway sẽ liên lạc với
gatekeeper khi có nhu cầu tìm kiếm thông tin này. Gatekeeper ngoài chức năng lưu trữ
thông tin định tuyến, phân giải cuộc gọi còn thực hiện các chức năng nâng cao như điều
khiển khả năng chấp nhận cuộc gọi mới (admission control), điều khiển băng thông
(bandwidth control) và quản lý vùng (zone management).
Yêu cầu cấu hình
• Cấu hình router 7200 như là gatekeeper
• Cấu hình hai router Saigon1, Saigon2 thực hiện cuộc gọi thông qua gatekeeper.
• Yêu cầu 2 gateway Saigon1 và Saigon2 thuộc cùng vùng (zone) Saigon trên
gatekeeper
Sơ đồ mạng
Hình 1
2. Hình 2
Thực hiện
A. Cấu hình ban đầu của hệ thống:
Gatekeeper#sh run
Building configuration...
Current configuration : 787 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Gatekeeper
!
ip cef
!
!
interface FastEthernet0/0
ip address 172.16.1.10 255.255.255.0
duplex auto
speed auto
!
3. !
control-plane
!
line con 0
exec-timeout 0 0
logging synchronous
stopbits 1
line aux 0
stopbits 1
line vty 0 4
!
end
Saigon1#sh run
Building configuration...
Current configuration : 982 bytes
!
version 12.3
!
hostname Saigon1
!
ip subnet-zero
ip cef
!
interface Ethernet0/0
ip address 172.16.1.1 255.255.255.0
half-duplex
!
!
ip http server
no ip http secure-server
ip classless
!
voice-port 1/0/0
!
voice-port 1/0/1
!
line con 0
line aux 0
line vty 0 4
privilege level 15
no login
!
end
Saigon2#sh run
Building configuration...
Current configuration : 958 bytes
!
4. version 12.3
hostname Saigon2
!
ip subnet-zero
ip cef
!
!
interface Ethernet0/0
ip address 172.16.1.2 255.255.255.0
half-duplex
!
!
ip http server
no ip http secure-server
ip classless
!
voice-port 1/0/0
!
voice-port 1/0/1
!
line con 0
line aux 0
line vty 0 4
login
!
!
end
B. Các bước thực hiện
Bước 1: Cấu hình gatekeeper
Vào mode cấu hình configuration
Gatekeeper#configure terminal
Gatekeeper(config)#gatekeeper <-- vào kiểu cấu hình gatekeeper
Gatekeeper (config−gk)#
Gatekeeper(config-gk)#zone local Saigon vnpro.org enable-intrazone
Định nghĩa vùng (zone) do gatekeeper quản lý
Cú pháp câu lệnh trên có dạng:
Gatekeeper (config−gk)#zone local gatekeeper−name domain−name
[ras−IP−address]
Gatekeeper-name: đặt tên cho vùng (zone) trên gatekeeper
Domain-name: tên domain đại diện
Trong câu lệnh trên, địa chỉ RAS là tùy chọn. Nếu bạn dùng tùy chọn này, khi
gatekeeper trả lời cho thông điệp tìm gatekeeper từ voice gateway, nó sẽ chỉ cho
các thiết bị đầu cuối dùng luôn địa chỉ này trong các truyền thông tương lai.
Trong trường hợp này dùng thêm từ khóa enable-intrazone cho phép định tuyến
các cuộc gọi giữa các gateway trong cùng một zone
5. Gatekeeper(config-gk)#zone prefix Saigon 1...
Khai báo số prefix của zone, trong trường hợp này vùng đại diện Saigon là vùng
có đầu số 1 và 3 số tiếp theo bất kỳ
Gatekeeper (config−gk)#no shutdown
Bật chức năng gatekeeper.
Bước 2: Cấu hình voice gateway
Saigon1#configure terminal
Saigon1(config)#gateway
Cấu hình cổng giao tiếp dùng H323 trên thiết bị gateway.
Saigon1(config)#interface ethernet 0/0
Saigon1(config−if)#h323−gateway voip interface
bật tính năng H323 voip trên cổng này
Saigon1(config−if)#h323−gateway voip h323−id Saigon1
Cú pháp câu lệnh có dạng:
Saigon1(config−if)#h323−gateway voip h323−id gateway−id
Gateway id trong trường hợp này là tên mà gateway chọn liên lạc với gatekeeper
(Saigon1)
Saigon1(config−if)#h323−gateway voip id Saigon ipaddr 172.16.1.10
Saigon1(config−if)#h323−gateway voip id gatekeeper−id {ipaddr
ip−address [port−number] | multicast}
Gatekeeper-id chính là tên của zone mà gatekeeper cấu hình thiết bị này thuộc về
Ip-address chính là địa chỉ IP của gatekeeper
Saigon1(config−if)#h323−gateway voip tech−prefix prefix
Cấu hình gateway để đăng ký gatekeeper với giá trị technology prefix, nếu dùng
technology prefix, technology prefix thể hiện loại dịch vụ sử dụng là voice hay
video.
Thực hiện cấu hình voice port, dial peers dạng pots, các VoIP dialer peer. Cấu hình
session target liên hệ đến gatekeeper dùng H.225 RAS.
Cấu hình các port FXS và số thoại tương ứng cho các port:
Saigon2(config)#dial-peer voice 1 pots
Saigon2(config-dial-peer)#
dial-peer voice 1 pots
6. destination-pattern 1121
port 1/0/0
!
dial-peer voice 2 pots
destination-pattern 1122
port 1/0/1
!
Cấu hình tham chiếu các số gọi có 4 ký số với 3 ký số đầu là 111 thì liên lạc với
gatekeeper
dial-peer voice 100 voip
destination-pattern 111.
session target ras
!
Cấu hình tương tự cho Saigon1 trở thành gateway (liên lạc thông qua cổng e0/0 trên thiết
bị) với số thoại liên hệ với cổng FXS là 1111 và 1112 đồng thời các tham chiếu đến số
gọi có 4 ký số với 3 ký số đầu là 112 thì liên lạc với gatekeeper.
Tiến hành quay số giữa 2 điện thoại 1111 và 1121 và kiểm tra thông tin
C. Kiểm tra cấu hình
Thực hiện cuộc gọi sẽ thành công,
Quan sát cấu hình dial-peer thực hiện trên các thiết bị gateway
Saigon2#sh dial-peer voice summary
dial-peer hunt 0
AD PRE PASS OUT
TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT
1 pots up up 1121 0 up 1/0/0
2 pots up up 1122 0 up 1/0/1
100 voip up up 111. 0 syst ras
Saigon2#
Saigon1#sh dial-peer voice summary
dial-peer hunt 0
AD PRE PASS OUT
TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT
1 pots up up 1111 0 up 1/0/0
2 pots up up 1112 0 up 1/0/1
100 voip up up 112. 0 syst ras
Saigon1#
7. Kiểm tra thông tin trên gatekeeper
Gatekeeper#sh gatekeeper status
Gatekeeper State: UP
Load Balancing: DISABLED
Flow Control: DISABLED
Zone Name: Saigon
Accounting: DISABLED
Endpoint Throttling: DISABLED
Security: DISABLED
Maximum Remote Bandwidth: unlimited
Current Remote Bandwidth: 0 kbps
Current Remote Bandwidth (w/ Alt GKs): 0 kbps
Hunt Scheme: Random
Gatekeeper#
8. Xem các thông tin đăng ký giữa gateway đến gatekeeper, có thể thấy các
thông tin số cấu hình trên các gateway được thông báo đến gatekeeper.
Gatekeeper#sh gatekeeper endpoints
GATEKEEPER ENDPOINT REGISTRATION
================================
CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags
--------------- ----- ------------- ----- --------- ---- -----
172.16.1.1 1720 172.16.1.1 54713 Saigon VOIP-GW
E164-ID: 1111
E164-ID: 1112
H323-ID: Saigon1
Voice Capacity Max.= Avail.= Current.= 1
172.16.1.2 1720 172.16.1.2 49153 Saigon VOIP-GW
H323-ID: Saigon2
E164-ID: 1121
E164-ID: 1122
Voice Capacity Max.= Avail.= Current.= 1
Total number of active registrations = 2
Gatekeeper#
Xem các thông tin cuộc gọi hiện thời
Gatekeeper#sh gatekeeper calls
Total number of active calls = 1.
GATEKEEPER CALL INFO
====================
LocalCallID Age(secs) BW
1-32559 1222 16(Kbps)
Endpt(s): Alias E.164Addr
src EP: Saigon2 1121
CallSignalAddr Port RASSignalAddr Port
172.16.1.2 1720 172.16.1.2 49153
Endpt(s): Alias E.164Addr
dst EP: Saigon1 1111
CallSignalAddr Port RASSignalAddr Port
172.16.1.1 1720 172.16.1.1 54713
Gatekeeper#
9. Xem thông tin cấu hình zone trên gatekeeper
Gatekeeper#sh gatekeeper zone prefix
ZONE PREFIX TABLE
=================
GK-NAME E164-PREFIX
------- -----------
Saigon 1...
Gatekeeper#
D. Cấu hình đầy đủ
Saigon1#sh run
Building configuration...
Current configuration : 982 bytes
!
version 12.3
!
hostname Saigon1
!
ip subnet-zero
ip cef
!
!
interface Ethernet0/0
ip address 172.16.1.1 255.255.255.0
half-duplex
h323-gateway voip interface
h323-gateway voip id Saigon ipaddr 172.16.1.10 1719
h323-gateway voip h323-id Saigon1
h323-gateway voip bind srcaddr 172.16.1.1
!
ip http server
no ip http secure-server
ip classless
!
voice-port 1/0/0
!
voice-port 1/0/1
!
dial-peer voice 1 pots
destination-pattern 1111
port 1/0/0
!
dial-peer voice 2 pots
destination-pattern 1112
port 1/0/1
!
dial-peer voice 100 voip
10. destination-pattern 112.
session target ras
!
gateway
!
line con 0
line aux 0
line vty 0 4
privilege level 15
no login
!
end
Saigon2#sh run
Building configuration...
Current configuration : 958 bytes
!
version 12.3
!
hostname Saigon2
ip subnet-zero
ip cef
!
interface Ethernet0/0
ip address 172.16.1.2 255.255.255.0
half-duplex
h323-gateway voip interface
h323-gateway voip id Saigon ipaddr 172.16.1.10 1719
h323-gateway voip h323-id Saigon2
h323-gateway voip bind srcaddr 172.16.1.2
!
ip http server
no ip http secure-server
ip classless
!
voice-port 1/0/0
!
voice-port 1/0/1
!
dial-peer voice 1 pots
destination-pattern 1121
port 1/0/0
!
dial-peer voice 2 pots
destination-pattern 1122
port 1/0/1
!
dial-peer voice 100 voip
destination-pattern 111.
session target ras
11. !
gateway
!
line con 0
line aux 0
line vty 0 4
login
!
!
end
Gatekeeper#sh run
Building configuration...
Current configuration : 787 bytes
!
version 12.4
!
hostname Gatekeeper
!
ip cef
!
!
interface FastEthernet0/0
ip address 172.16.1.10 255.255.255.0
duplex auto
speed auto
!
!
no ip http server
no ip http secure-server
!
!
control-plane
!
gatekeeper
zone local Saigon vnpro.org enable-intrazone
zone prefix Saigon 1...
no shutdown
!
line con 0
exec-timeout 0 0
logging synchronous
stopbits 1
line aux 0
stopbits 1
line vty 0 4
!
end
