Join us for an introductory webinar on VoIP and learn:
- The fundamental principles of VoIP including RTP and SIP
- What voice metrics to measure and why they matter
- The different methods to monitor and troubleshoot VoIP
This document outlines a project proposal to compare VoIP systems through simulation. The objectives are to analyze VoIP voice quality using different codecs, equipment, phone frequencies, and bandwidth requirements. An overview of VoIP technology is provided along with descriptions of common VoIP protocols like SIP, H.323, and MGCP. Factors that impact VoIP quality like packet loss, delay, and jitter are explained. Popular codecs like G.711, G.729, and GSM are described along with their bandwidth requirements. A timeline is given for the project with submission deadlines throughout April to July.
The document discusses technologies relevant to voice over IP (VoIP) applications including voice processing modules, codecs, signaling protocols, transport protocols, and network traversal techniques. It also covers business models, community aspects, and opportunities in premium services and bridging across technologies and communities. Emerging areas discussed include VoIP on mobile networks and the roles of portals, social networks, and device vendors.
VoIP allows transmission of voice calls over IP networks like the internet. It works by encoding voice input and transmitting it as digital audio packets rather than traditional circuit-switched telephone networks. Key challenges include latency, jitter, bandwidth, packet loss, reliability, and security. Standards like H.323, SIP, and MGCP help address these and ensure interoperability between VoIP systems from different vendors. Components include servers, end-point devices, media gateways, and the IP network itself.
The document introduces VoIP (Voice over IP) concepts. It discusses digitization of audio, real-time compression/encoding, transport over UDP, and problems with UDP like packet loss and jitter. It also covers protocols like SIP for signaling, SDP for session description, and RTP for media transport. Key VoIP services that can be implemented with SIP are discussed, like call transfer and voicemail.
Practical Fundamentals of Voice over IP (VoIP) for Engineers and TechniciansLiving Online
In the past five years, technologies have converged to such an extent that one can transmit voice, fax and video over the same internet protocol network that one uses for data. This workshop examines Voice over IP (VoIP) technologies and provides you with the skills to competently implement a VoIP network for your organisation. Numerous case studies and exercises throughout the course ensure that you get a good grasp on the technologies used. Solid practical advice is given on application, implementation and most importantly troubleshooting these systems.
MORE INFORMATION: http://www.idc-online.com/content/practical-fundamentals-voice-over-ip-voip-engineers-and-technicians-3
A VoIP gateway acts as an interface between a public switched telephone network (PSTN) and an IP network, converting voice and fax calls between the two in real time. Key functions include voice and fax compression/decompression, packetization, and call routing. There are analog gateways for connecting PSTN lines to VoIP systems and digital gateways for connecting PBX systems. When selecting a gateway, factors to consider include call load, supported protocols and compatibility, and cost.
This document provides an overview of Voice over IP (VoIP) technology. It explains that VoIP allows voice calls to be made over an IP network like the internet rather than the traditional public switched telephone network. It describes how VoIP works by converting voice signals to digital form and transmitting them in packets over the network. It also outlines the key components of a VoIP system including protocols, gateways, codecs and the network architecture. Additionally, it discusses benefits of VoIP like reduced communication costs and better utilization of bandwidth.
VoIP (Voice Over IP) allows users to make phone calls using an Internet connection rather than a traditional phone line. It works by converting voice signals into digital data packets which are transmitted over the Internet or other IP-based networks. Common protocols used for VoIP include UDP, RTP, and SIP. While VoIP provides advantages like lower costs, it also faces challenges of packet loss, latency, jitter, and firewall restrictions that can impact call quality.
Efficient Telecommunication Infrastructure with Internet Telephony (VoIP)Thomas Siegers
This document provides an overview of efficient telecommunication infrastructure using Internet telephony (VoIP). It discusses basics of telephony and networking, protocols like SIP and RTP, VoIP hardware, service providers, and examples of integrating VoIP into networks and PBX systems. Skype and issues with it are also covered. The presentation aims to explain how VoIP works and scenarios for implementing efficient VoIP solutions.
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.
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.
Voice over Internet Protocol (VoIP) is a technology that allows users to make voice calls over an Internet Protocol (IP) network like the Internet instead of traditional telephone lines. It converts analog voice signals to digital data that can be sent in IP packets over broadband connections. Standards organizations like the IETF and ITU work to standardize VoIP protocols to ensure interoperability. Common VoIP protocols include SIP, H.323, and MGCP. VoIP allows for phone-to-phone calls over the Internet as well as phone-to-PC and PC-to-phone calls. Services like Vonage and Skype use VoIP to allow users to make inexpensive calls over the Internet.
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
Voice over IP (VoIP) allows voice traffic to be carried over an IP data network at lower bandwidth than traditional telephone networks. It provides benefits such as lower communication costs, convergence of voice and data infrastructure, and new multimedia applications. However, VoIP also faces issues including delay, congestion, jitter, packet loss, bandwidth limitations, echo, interoperability between different systems, and ensuring scalability. The two main VoIP protocols are the Session Initiation Protocol (SIP) and H.323. VoIP adoption is growing due to the increasing use of IP networks, and it provides opportunities for lower telephone costs and innovative services. However, challenges remain regarding quality of service, interoperability, and developing carrier-grade
Voice Over Internet Protocol (VoIP) allows both voice and data calls to be made over an Internet Protocol (IP) network. It integrates telephone services and computer applications, providing a more flexible and cost-effective communication system compared to traditional telephone networks. The document describes a student project to implement VoIP using software on a local area network, which successfully transmitted sound between a phone, computer, and between computers. It discusses expanding the system to support multiple simultaneous users over the Internet.
VoIP allows analog voice signals to be transmitted over the internet by converting voice data into digital packets. At the sender, the analog voice input is converted to digital data using codecs and loaded into IP packets according to protocols. These packets are transmitted over the internet and received, where they are merged back into a binary data stream and converted to analog voice using codecs. Common protocols used include SIP for signaling and RTP within UDP/IP for transmitting voice data packets in real-time. VoIP provides advantages over PSTN like lower costs and increased functionality like conferencing and simultaneous voice and data transmission.
This document provides an overview of VoIP techniques and challenges. It begins with an introduction to VoIP and what it is. It then discusses quality of service (QoS) and the importance of providing prioritized delivery services for applications like VoIP. The main challenges for VoIP are then outlined as system capacity/available bandwidth, packet loss, delay/network latency, jitter, echo, and security. Specific causes and issues are described for each challenge. The conclusion is that while VoIP provides a cheaper alternative to PSTN, it has lower quality of service due to these challenges, and the internet is not perfectly designed to carry voice.
The document discusses Jingle, an open standard protocol for real-time communication like voice and video calls over the XMPP protocol. Jingle allows for peer-to-peer connections using techniques like STUN and ICE to traverse NATs and firewalls, with the ability to fallback to using media servers. It describes the Jingle protocol which negotiates sessions and transports in three steps. Open source libraries and servers like libjingle and Openfire support Jingle implementations.
This document discusses quality of service (QoS) requirements for voice over IP (VoIP) and how QoS can be implemented in packet switched networks to address issues like jitter, latency, bandwidth congestion, and packet loss that can negatively impact call quality. It explains that QoS aims to guarantee a certain level of performance for applications like VoIP through techniques like traffic classification, marking, and queuing. The document also provides recommendations for applying QoS on the network edge, core, and internet exchange points to help improve end-to-end call quality.
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.
1. The document introduces VoIP concepts and presents Asterisk as a free and open source PBX software solution that is well-suited for implementing VoIP networks in developing regions.
2. It discusses challenges in developing regions like lack of technical knowledge and affordable infrastructure, and how VoIP solutions like Asterisk can help address these issues by providing flexibility.
3. The document provides an overview of topics covered like basic VoIP concepts, how to set up an Asterisk PBX, equipment options, and presents a case study of introducing VoIP services.
a seminar paper presentation .this will help you know about voice transmission over the internet protocol's.as in Skype, watts app. it also give an idea about old technology. thanks. if any mistakes ,and add any updates and share with me .on about this slide
VoIP (Voice over Internet Protocol) allows users to make phone calls using an internet connection instead of a regular phone line. It emerged as an alternative to the traditional Public Switched Telephone Network. Some key points:
- The first VoIP software was introduced in 1995 and used H.323 protocol over home PCs.
- There are several VoIP architectures including PC-to-PC, phone-to-phone via internet, and connections between the internet and PSTN.
- Popular VoIP protocols are SIP, IAX, H.323, and IMS.
- VoIP provides advantages like lower costs, integration of voice and data, and video conferencing capabilities. Disadv
Avaya VoIP on Cisco Best Practices by PacketBasePacketBase, Inc.
The document provides an overview of Avaya IP communications and best practices for interoperability with Cisco networks. It discusses key considerations for quality of service including recommended delay, jitter and packet loss thresholds. It also provides guidance on general QoS approaches, IP phone deployment, VLAN configuration, QoS settings for Cisco switches, and best practices for WAN connectivity.
VoIP, or Voice over Internet Protocol, is a technology that allows routing of voice data through IP-based networks rather than traditional circuit-switched transmission lines. This allows voice transmission over a packet-switched network and provides benefits like cost reduction, toll bypassing, common network infrastructure, and simplified routing administration. VoIP integration with other business tools also allows for unified messaging through voice, email, and fax via the internet on both computers and mobile devices using IP networks. Common VoIP setups involve VoIP phones, analog phones connected to VoIP adapters, and softphones that allow making calls directly from a computer.
This document provides an overview of open source PBX software called Asterisk. It discusses VoIP technologies including codecs, protocols and PBX features. It also outlines how to install, configure and use Asterisk to set up a PBX system with channels, phones, IVRs and billing integration. Hardware requirements and options for interfaces are presented along with examples of configuration files. The document demonstrates how to register softphones and test calling between Asterisk and other VoIP systems.
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.
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 (Voice Over IP) allows users to make phone calls using an Internet connection rather than a traditional phone line. It works by converting voice signals into digital data packets which are transmitted over the Internet or other IP-based networks. Common protocols used for VoIP include UDP, RTP, and SIP. While VoIP provides advantages like lower costs, it also faces challenges of packet loss, latency, jitter, and firewall restrictions that can impact call quality.
Efficient Telecommunication Infrastructure with Internet Telephony (VoIP)Thomas Siegers
This document provides an overview of efficient telecommunication infrastructure using Internet telephony (VoIP). It discusses basics of telephony and networking, protocols like SIP and RTP, VoIP hardware, service providers, and examples of integrating VoIP into networks and PBX systems. Skype and issues with it are also covered. The presentation aims to explain how VoIP works and scenarios for implementing efficient VoIP solutions.
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.
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.
Voice over Internet Protocol (VoIP) is a technology that allows users to make voice calls over an Internet Protocol (IP) network like the Internet instead of traditional telephone lines. It converts analog voice signals to digital data that can be sent in IP packets over broadband connections. Standards organizations like the IETF and ITU work to standardize VoIP protocols to ensure interoperability. Common VoIP protocols include SIP, H.323, and MGCP. VoIP allows for phone-to-phone calls over the Internet as well as phone-to-PC and PC-to-phone calls. Services like Vonage and Skype use VoIP to allow users to make inexpensive calls over the Internet.
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
Voice over IP (VoIP) allows voice traffic to be carried over an IP data network at lower bandwidth than traditional telephone networks. It provides benefits such as lower communication costs, convergence of voice and data infrastructure, and new multimedia applications. However, VoIP also faces issues including delay, congestion, jitter, packet loss, bandwidth limitations, echo, interoperability between different systems, and ensuring scalability. The two main VoIP protocols are the Session Initiation Protocol (SIP) and H.323. VoIP adoption is growing due to the increasing use of IP networks, and it provides opportunities for lower telephone costs and innovative services. However, challenges remain regarding quality of service, interoperability, and developing carrier-grade
Voice Over Internet Protocol (VoIP) allows both voice and data calls to be made over an Internet Protocol (IP) network. It integrates telephone services and computer applications, providing a more flexible and cost-effective communication system compared to traditional telephone networks. The document describes a student project to implement VoIP using software on a local area network, which successfully transmitted sound between a phone, computer, and between computers. It discusses expanding the system to support multiple simultaneous users over the Internet.
VoIP allows analog voice signals to be transmitted over the internet by converting voice data into digital packets. At the sender, the analog voice input is converted to digital data using codecs and loaded into IP packets according to protocols. These packets are transmitted over the internet and received, where they are merged back into a binary data stream and converted to analog voice using codecs. Common protocols used include SIP for signaling and RTP within UDP/IP for transmitting voice data packets in real-time. VoIP provides advantages over PSTN like lower costs and increased functionality like conferencing and simultaneous voice and data transmission.
This document provides an overview of VoIP techniques and challenges. It begins with an introduction to VoIP and what it is. It then discusses quality of service (QoS) and the importance of providing prioritized delivery services for applications like VoIP. The main challenges for VoIP are then outlined as system capacity/available bandwidth, packet loss, delay/network latency, jitter, echo, and security. Specific causes and issues are described for each challenge. The conclusion is that while VoIP provides a cheaper alternative to PSTN, it has lower quality of service due to these challenges, and the internet is not perfectly designed to carry voice.
The document discusses Jingle, an open standard protocol for real-time communication like voice and video calls over the XMPP protocol. Jingle allows for peer-to-peer connections using techniques like STUN and ICE to traverse NATs and firewalls, with the ability to fallback to using media servers. It describes the Jingle protocol which negotiates sessions and transports in three steps. Open source libraries and servers like libjingle and Openfire support Jingle implementations.
This document discusses quality of service (QoS) requirements for voice over IP (VoIP) and how QoS can be implemented in packet switched networks to address issues like jitter, latency, bandwidth congestion, and packet loss that can negatively impact call quality. It explains that QoS aims to guarantee a certain level of performance for applications like VoIP through techniques like traffic classification, marking, and queuing. The document also provides recommendations for applying QoS on the network edge, core, and internet exchange points to help improve end-to-end call quality.
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.
1. The document introduces VoIP concepts and presents Asterisk as a free and open source PBX software solution that is well-suited for implementing VoIP networks in developing regions.
2. It discusses challenges in developing regions like lack of technical knowledge and affordable infrastructure, and how VoIP solutions like Asterisk can help address these issues by providing flexibility.
3. The document provides an overview of topics covered like basic VoIP concepts, how to set up an Asterisk PBX, equipment options, and presents a case study of introducing VoIP services.
a seminar paper presentation .this will help you know about voice transmission over the internet protocol's.as in Skype, watts app. it also give an idea about old technology. thanks. if any mistakes ,and add any updates and share with me .on about this slide
VoIP (Voice over Internet Protocol) allows users to make phone calls using an internet connection instead of a regular phone line. It emerged as an alternative to the traditional Public Switched Telephone Network. Some key points:
- The first VoIP software was introduced in 1995 and used H.323 protocol over home PCs.
- There are several VoIP architectures including PC-to-PC, phone-to-phone via internet, and connections between the internet and PSTN.
- Popular VoIP protocols are SIP, IAX, H.323, and IMS.
- VoIP provides advantages like lower costs, integration of voice and data, and video conferencing capabilities. Disadv
Avaya VoIP on Cisco Best Practices by PacketBasePacketBase, Inc.
The document provides an overview of Avaya IP communications and best practices for interoperability with Cisco networks. It discusses key considerations for quality of service including recommended delay, jitter and packet loss thresholds. It also provides guidance on general QoS approaches, IP phone deployment, VLAN configuration, QoS settings for Cisco switches, and best practices for WAN connectivity.
VoIP, or Voice over Internet Protocol, is a technology that allows routing of voice data through IP-based networks rather than traditional circuit-switched transmission lines. This allows voice transmission over a packet-switched network and provides benefits like cost reduction, toll bypassing, common network infrastructure, and simplified routing administration. VoIP integration with other business tools also allows for unified messaging through voice, email, and fax via the internet on both computers and mobile devices using IP networks. Common VoIP setups involve VoIP phones, analog phones connected to VoIP adapters, and softphones that allow making calls directly from a computer.
This document provides an overview of open source PBX software called Asterisk. It discusses VoIP technologies including codecs, protocols and PBX features. It also outlines how to install, configure and use Asterisk to set up a PBX system with channels, phones, IVRs and billing integration. Hardware requirements and options for interfaces are presented along with examples of configuration files. The document demonstrates how to register softphones and test calling between Asterisk and other VoIP systems.
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.
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 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.
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
02 asterisk - the future of telecommunicationsTran Thanh
Asterisk is an open-source private branch exchange (PBX) system that can be used to build voice over IP (VoIP) networks and systems. It allows users to reproduce standard PBX features and interact IP-based networks. Asterisk is hardware independent and can run on various operating systems. It provides implementations of basic PBX functionality and integrates with third-party telephony hardware and software.
Telephony Service Development on Asterisk PlatformHamid Fadishei
Asterisk is a major role player in the open source telecom world. In this workshop, participants will follow a step-by-step case study towards getting familiar with IVR service development on Asterisk platform using PHP programming language and AGI technology. The case study itself is a simple weather forecasting IVR service.
This document discusses building a SIP softswitch using Asterisk and Asterisk-Java. It describes using Asterisk to handle SIP signaling, media processing, and interfacing with a Java routing application. The Java application directs call routing and interfaces with Asterisk via FastAGI and AMI. Key issues addressed include having call legs survive independently and supporting early media. Patches are used to enable call bridging and configurable early media handling in app_dial.
This document provides an introduction to installing and configuring the Asterisk PBX software. It begins with an agenda that covers installing Asterisk, the basics of how Asterisk works, configuring telephony hardware, basic configuration files like sip.conf and extensions.conf, and advanced topics like voicemail, conferencing and scaling. It then discusses downloading and compiling the required components like Zaptel and Libpri as well as configuring SIP endpoints. The document provides an overview of Asterisk's architecture and components and how to structure dial plans using contexts, extensions, applications and variables.
Nowadays VoIP technologies have taken the upper hand offering many advantages compared to the traditional telephone network, but what are the security risks involved when voice and data networks come together. In this presentation, we will identify and evaluate these different security risks and their countermeasures both from a defensive as offensive position.
The document discusses using SIP (Session Initiation Protocol) for applications beyond just voice over IP (VoIP) communication. It describes how SIP was designed to be extensible with different media streams beyond just audio and video. It provides examples of using SIP for presence, instant messaging, file transfer and desktop sharing using protocols like SIMPLE, MSRP. It also discusses building collaboration tools with SIP, including features like conferencing, chat rooms and screen sharing. The document promotes moving beyond just VoIP with SIP to enable new communication and collaboration applications.
The document discusses vulnerabilities and attacks against Voice over IP (VoIP) systems. It begins with an introduction to VoIP architecture, components, and protocols. It then covers vulnerabilities and common attack vectors against VoIP, such as identity spoofing, eavesdropping, password cracking, and denial of service attacks. The document demonstrates some example attacks and outlines tools that can be used for scanning, attacking, and testing the security of VoIP systems. It concludes with recommendations for countermeasures like firewalls, encryption, and network hardening to better secure VoIP infrastructures.
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 Art of VoIP Hacking - Defcon 23 WorkshopFatih Ozavci
VoIP attacks have evolved, and they are targeting Unified Communications (UC), commercial services, hosted environment and call centres using major vendor and protocol vulnerabilities. This workshop is designed to demonstrate these cutting edge VoIP attacks, and improve the VoIP skills of the incident response teams, penetration testers and network engineers. Signalling protocols are the centre of UC environments, but also susceptible to IP spoofing, trust issues, call spoofing, authentication bypass and invalid signalling flows. They can be hacked with legacy techniques, but a set of new attacks will be demonstrated in this workshop. This workshop includes basic attack types for UC infrastructure, advanced attacks to the SIP and Skinny protocol weaknesses, network infrastructure attacks, value added services analysis, Cdr/Log/Billing analysis and Viproy use to analyse signalling services using novel techniques. Also the well-known attacks to the network infrastructure will be combined with the current VoIP vulnerabilities to test the target workshop network. Attacking VoIP services requires limited knowledge today with the Viproy Penetration Testing Kit (written by Fatih). It has a dozen modules to test trust hacking issues, information collected from SIP and Skinny services, gaining unauthorised access, call redirection, call spoofing, brute-forcing VoIP accounts, Cisco CUCDM exploitation and debugging services using as MITM. Furthermore, Viproy provides these attack modules in the Metasploit Framework environment with full integration. The workshop contains live demonstration of practical VoIP attacks and usage of the Viproy modules.
In this hands-on workshop, attendees will learn about basic attack types for UC infrastructure, advanced attacks to the SIP protocol weaknesses, Cisco Skinny protocol hacking, hacking Cisco CUCDM and CUCM servers, network infrastructure attacks, value added services analysis, Cdr/Log/Billing analysis and Viproy VoIP pen-test kit to analyse VoIP services using novel techniques. New CDP, CUCDM and Cisco Skinny modules and techniques of Viproy will be demonstrated in the workshop as well.
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.
The document discusses Voice over Internet Protocol (VoIP) as an alternative to traditional telephone lines for providing reference services in libraries. VoIP allows for voice communication over broadband internet and has low requirements and costs for libraries. It offers advantages like low costs, extra functionality, and allowing voice chat with customers. Potential disadvantages include needing training, relying on electricity, and having spotty emergency services. The document argues that VoIP could greatly enhance reference interviews by allowing clearer and quicker communication compared to text.
Maroc telecom - New scope, new rating - Buy (vs. Hold) – Target price: MAD 11...mehdiammouri
1) CFG Research upgraded their rating of Maroc Telecom from Hold to Buy following the company's acquisition of Moov assets, which will increase revenues and earnings growth potential.
2) Domestic mobile revenues are expected to decline in the near term due to continued competitive pressure, while fixed line and broadband revenues are forecasted to grow with an improving regulatory environment.
3) The acquisition of Moov assets expands Maroc Telecom's operations into new markets in sub-Saharan Africa with lower penetration rates and growth potential, but also higher risks due to varying competitive landscapes and political risks across countries.
Telephone Wreckers tells you all about Asterisk phone systems - the benefits, features and what product you'll need to build your own custom IP phone system.
Asterisk is open-source software that allows the creation of voice over IP (VoIP) systems. It supports protocols like SIP and H.323 and can integrate with hardware from companies like Digium to connect to traditional phone networks. Asterisk has a modular design that allows it to be used in various configurations, from a full PBX to smaller applications like conferencing bridges. It has a large community for support and runs on Linux and other operating systems.
This document discusses best practices for deploying WebRTC to replace or augment existing SIP-based phone systems. It covers choosing appropriate codecs to balance bandwidth usage and call quality for different use cases. It also addresses WebRTC-specific considerations like ICE, DTLS, and asymmetric call patterns. Performance metrics are provided from test calls using different codecs on an Asterisk server. The presentation includes diagrams of common WebRTC deployment architectures and links to live demos.
Getting the Best Out Of WebRTC - Astricon 2014Dan Jenkins
This document discusses best practices for deploying WebRTC to replace or augment existing SIP-based phone systems. It covers choosing appropriate codecs to balance bandwidth usage and call quality for different use cases. It also addresses WebRTC-specific considerations like ICE, DTLS, and asymmetric call patterns. Performance metrics are provided from test calls using different codecs on an Asterisk server. The presentation includes diagrams of common WebRTC deployment architectures and links to live demos.
MyPBX is a standalone embedded hybrid PBX for small businesses and remote branch offices that offers both VoIP and traditional phone system solutions. It supports up to 100 users with 22 concurrent calls and has features such as voicemail, conferencing, call forwarding, and call recording. The compact device has Ethernet, analog, and E1/T1 ports and supports codecs like G.711, G.722, and G.729.
MyPBX E1 is a standalone embedded hybrid PBX for small businesses and remote branch offices that offers both VoIP and traditional telephony solutions. It supports up to 100 users and 22 concurrent calls, has E1/T1/J1 ports, 8 analog ports, and features like voicemail, conferencing, and call routing. The compact device has ample processing power and memory as well as standard and wideband codecs for high quality calls.
MyPBX is a standalone embedded hybrid PBX for small businesses and remote branch offices that offers a combination of VoIP and legacy telecom equipment. It provides features such as auto provisioning, call forwarding, conferencing, voicemail, and a web-based interface. The document provides specifications and contact information for obtaining demonstrations or further information.
This document provides an overview and objectives for an E-Learning training course on building a complete PBX with Asterisk. The training will cover understanding and installing Asterisk, building a simple PBX with SIP phones and SIP trunks, configuring features like dialplans, IVRs and voicemail, and integrating applications. By the end of the course, students will have hands-on experience building their own basic Asterisk system. Various chapters include objectives, presentations on topics, and instructions for labs to gain practical experience configuring Asterisk.
The document discusses the open source Asterisk PBX software. It provides an overview of Asterisk including that it was created in 1999 as a free and open source alternative to expensive proprietary PBX systems. Asterisk allows users to build their own software-based phone systems using inexpensive hardware and can provide many of the same features as traditional PBXs through its flexible architecture and extensive capabilities. The document outlines some of Asterisk's main functionalities and how it works as well as hardware that can be used with it.
The document discusses open source VoIP and Asterisk. It summarizes that Asterisk is the most popular open source IP PBX option. While some enterprises were hesitant to adopt open source for phone systems due to concerns over support and compatibility, Asterisk has proven viable for large implementations. Asterisk can integrate analog and IP phones to replace traditional PBXs in a cost-effective manner.
Asterisk is an open-source private branch exchange (PBX) system that supports voice over IP (VoIP) technology. It runs on many operating systems and supports several protocols. Asterisk can integrate with both IP phones and traditional phone systems. It offers a wide range of features including voicemail, conferencing, interactive voice response, and contact centers. Asterisk works with various hardware and software to provide both on-premise and cloud-based phone systems for businesses of all sizes.
*astTECS IP PBX is the right choice for every business sizes. *astTECS IP PBX is an open source communication platform which provides all the advanced features at affordable price. The solution is easy to manage, future ready solution , increase productivity and can integrate seamlessly with existing telecom infrastructure. *astTECS IP PBX is TEC certified by DoT, Govt of India and we are first open source company to get the certification. Open Source Astrisk Platform – No licensing and vendor lock in . Cost effective -save upto 80% telecom expenses. Unlimited CTI integration - Seamless integration with third party Applications. Advanced and real time dashboards and statistics.
Asterisk is an Open Source PBX - but how does it support larger installations? Can you scale it up to thousands of users, with hundreds of simultaneous calls? What about failover, backups and the famous blinking lamps? Olle Johansson goes through various models and describes where some of his current projects with strange names - Pinefrog, Pinana, Pinetree and Bufo fits into this picture.
This document summarizes Cisco small business solutions and products, including:
1. Kennedy Communications provides Cisco phone systems, computer networking, and post-sale support for small businesses across the Southeast.
2. Cisco offers purpose-built networking, security, storage, and voice/video solutions for small businesses, including the Cisco Small Business Pro Series and SBCS unified communications appliances.
3. New Cisco products highlighted include the SPA500 series phones, ESW500 switches, SA500 security appliances, network video cameras, and network storage devices.
The document introduces the *astTECS IP PBX system. It is a true IP PBX that offers flexibility, scalability, advanced features, and easy integration. The system uses a web-based interface for simple configuration and management without vendor locking. It can connect via various interfaces to support IP phones, analog phones, and third party applications.
The document introduces the *astTECS IP PBX system. It is a true IP PBX that offers flexibility, scalability, advanced features, and easy integration. The system uses a web-based interface for simple configuration and management without vendor locking. It can connect via various interfaces to phones, cameras, gateways and more. The system provides call features, conferencing, voicemail, and can integrate with third party apps.
The document introduces the *astTECS IP PBX system. It is a true IP PBX that offers flexibility, scalability, advanced features, and easy integration. The system uses a web-based interface for simple configuration and management without vendor locking. It can connect via various interfaces to support IP phones, analog phones, and third party applications.
The MyPBX SOHO is a standalone IP-PBX for small businesses with up to 32 users that offers flexibility through integration of ISDN, PSTN lines, and VoIP trunks. It provides enterprise-class communication features to reduce communication costs by taking advantage of VoIP technology while ensuring reliability. It has a customizable combination of FXO, FXS, and BRI modules and is compatible with a wide range of IP phones through a web-based interface.
MyPBX Standard is an embedded hybrid PBX designed for small businesses with up to 100 users. It offers flexibility by integrating ISDN, PSTN lines, and VoIP/GSM/UMTS trunks, allowing businesses to reduce communication costs by taking advantage of VoIP technology while maintaining reliability through traditional phone lines. It provides enterprise-class communication features through an easy to use web interface and supports a wide range of IP phone models.
From Efficiency to Innovation: Transforming Business Value through Gen AISameer Verma
The world of Al is undergoing a metamorphosis. Traditional Al, programmed for specific tasks like playing chess, is being eclipsed by the new era of learning Al. This new breed can adapt, analyze data, and even create content. This shift is a game-changer for enterprises. Repetitive tasks can be automated, vast datasets can be analyzed for insights, and even entirely new products can be Al-powered. But the workforce needs to adapt too. Collaboration with Al tools will be key, requiring new skillsets like critical thinking and problem-solving. Generative Al, with its ability to craft images, music, and even code, holds immense promise. However, current offerings are in their infancy they can be impressive but prone to stumbles and biases.
The future of business is a partnership with Al. Businesses must carefully assess current tools and invest in human-Al collaboration and continuous learning. This will be the key to navigating the exciting, but uncertain path ahead. Eventually, we must not lose sight of the true purpose of an enterprise to provide value to the consumers, in order to improve their lives, and to do so responsibly, and in a sustainable way that provides acceptable returns to stakeholders.
A Framework for Information Access in Rural and Remote CommunitiesSameer Verma
Access to information is predicated on the access to a digital infrastructure. However, access to electricity and the Internet remain elusive for a significant percentage of the world's population, let alone a sustainable access in one’s local language, local context, and relating to local culture. This paper examines the issues of resource constraints, and proposes a framework to classify them. It then proceeds to utilize this framework to look at three different case studies of implementations of offline Internet access in Madagascar, Jamaica and India.
Presented at IEEE ISTAS 2016. http://istas2016.org
This document describes the XOVis learning analytics and visualization tool. XOVis collects metadata from students' work on their laptops to provide insights into learning and engagement. Student work is stored locally and then synced across schools and to the cloud using CouchDB and eventual consistency. This allows analytics even when internet is unavailable. XOVis processing and reporting is done both at local school appliances and in the cloud. The goal is to help educators better understand learning through visualized analytics on student computer usage.
Juju, LXC, OpenStack: Fun with Private CloudsSameer Verma
Description: Private clouds fill an interesting space in the cloud roadmap. They can provide a scalable, reliable, fault-tolerant cloud platform on your own infrastructure, and can be balanced with public cloud offerings. We will look at three technologies. OpenStack is a cloud operating system that controls large pools of compute, storage, and networking resources throughout a datacenter, all managed through a dashboard that gives administrators control while empowering their users to provision resources through a web interface. Juju, a cloud orchestration platform from Ubuntu, enables you to build entire environments in the cloud with only a few commands on public clouds like Amazon Web Services and HP Cloud, to private clouds built on OpenStack. LXC is the userspace control package for Linux Containers, a lightweight virtual system mechanism sometimes described as “chroot on steroids”. LXC builds up from chroot to implement complete virtual systems, adding resource management and isolation mechanisms to Linux’s existing process management infrastructure. How cool would it be, to walk around with a private cloud on your laptop?
"Computer, end program": Virtualization and the CloudSameer Verma
One does not simply explain "cloud". A continuum from virtual machines to the cloud, with a Star Trek bias. Holodeck, virtual machines, hypervisors, pulbic cloud, private cloud, hybrid cloud, VirtualBox, Ubuntu, OpenStack, and finally, Make it so!
Creativity and Innovation with One Laptop per ChildSameer Verma
How the One Laptop per Child project comes up with creative and innovative solutions to challenging problems by changing the constraints to the problems.
The document discusses the One Laptop per Child (OLPC) initiative which aims to provide low-cost and rugged laptops to empower education for children in developing areas of the world. It has distributed over 3 million laptops to children in over 40 countries speaking over 30 languages. The laptops use the Sugar interface and are designed for collaborative, joyful learning through activities like TurtleArt, Scratch, and measuring. OLPC has implementations in specific areas described like Nigeria, Thailand, India, Mongolia, Ethiopia, and more.
The Joy of Z Axis: Creativity and Innovation through 3D PrintingSameer Verma
Presentation on creativity and innovation through 3D printing. Featuring the Printrbot Jr. V2 at the College of Business, San Francisco State University.
One Laptop per Child and Sugar: Collaborative, Joyful and Self-empowered Lear...Sameer Verma
The One Laptop Per Child (OLPC) project has had several beginnings. The idea has roots in the 60s. It gained momentum in the last 15 years. OLPC released the idea to the world in 2005, and its first product in 2007. A lot has changed since then. We'll look at an update on the projects, learning through robotics, assessment through learning analytics, offline mirco-clouds, HTML5 apps, Sugar on tablets and Raspberry Pi, and other new initiatives. In a world of cheap, Android-driven tablets, how does the idea of OLPC fit? What role does the Sugar learning platform continue to play inside and outside of OLPC? Help us grow the initiatives so that children of the world may continue to have a chance at collaborative, joyful, and self-empowered learning.
Education and Social Inclusion through InformationSameer Verma
The document discusses the One Laptop Per Child (OLPC) organization, which aims to empower children worldwide through education. Its mission is to provide each child with a low-cost, rugged laptop to support collaborative and self-directed learning. OLPC has distributed over 3 million laptops to children in over 40 countries. The document outlines OLPC's educational approach and principles, technical specifications for its XO laptop, and its software platform and learning content. It also describes OLPC's architecture which utilizes cloud, on-site micro-cloud, and individual devices to enable learning even without internet connectivity.
Data by itself is simply a collection of numbers. It only becomes meaningful when we weave it through context. A context of relevance that creates information - provides insight, creates solutions and solves problems. The Web gives us a fabric of connectedness, but if the data isn't substantiated semantically, the information we create isn't very useful. By building effective web assets using platforms like Drupal, we build ways to solve problems across the spectrum from local to global. We not only build the Web the way it was meant to be, but we also build it to support a commons across community, enterprise and government for generations to come.
An introduction to virtualization as a concept, its implementation in VirtualBox and an extension into an OpenStack private cloud. Done at SF State University. See more at http://commons.sfsu.edu/virtualization-and-cloud
Social Justice and Equity through InformationSameer Verma
This document summarizes a presentation by Sameer Verma on social justice and equity through information. It discusses how free and open source software can help increase access to information for underserved communities and reduce the digital divide. It provides examples of how One Laptop Per Child is working to provide low-cost laptops and educational resources to children in over 40 countries worldwide, especially in rural areas lacking technology and infrastructure. The presentation emphasizes using technology and information to empower communities and further social justice and equity goals.
Social Justice and Equity through InformationSameer Verma
This document summarizes a presentation about social justice and equity through information and technology. It discusses how free and open source software can help increase access to information globally. It provides examples of the One Laptop Per Child (OLPC) initiative that aims to provide low-cost laptops to children in developing countries around the world. Specific examples of OLPC programs in countries like India, Jamaica, Afghanistan and partnerships with San Francisco State University are mentioned. The document advocates that technologies like OLPC can help more of the world gain access to education and information.
Explore the growing trend of payroll outsourcing in the UK with key 2025 statistics, market insights, and benefits for accounting firms. This infographic highlights why more firms are turning to outsourced payroll services for UK businesses to boost compliance, cut costs, and streamline operations. Discover how QXAS can help your firm stay ahead.
for more details visit:- https://qxaccounting.com/uk/service/payroll-outsourcing/
Diagrams are key to architectural work, aligning teams and guiding business decisions. This session covers best practices for transforming text into clear flowcharts using standard components and professional styling. Learn to create, customize, and reuse high-quality diagrams with tools like Miro, Lucidchart, ... Join us for hands-on learning and elevate your diagramming skills!
NewBase 05 May 2025 Energy News issue - 1785 by Khaled Al Awadi_compressed.pdfKhaled Al Awadi
Greetings,
Hawk Energy is pleased to share with you its latest energy news from NewBase Energy
as per attached file NewBase 05 May 2025 Energy News issue - 1785 by Khaled Al Awadi
Regards.
Founder & Senior Editor NewBase Energy
Khaled M Al Awadi, Energy ConsultantGreetings,
Hawk Energy is pleased to share with you its latest energy news from NewBase Energy
as per attached file NewBase 05 May 2025 Energy News issue - 1785 by Khaled Al Awadi
Regards.
Founder & Senior Editor NewBase Energy
Khaled M Al Awadi, Energy ConsultantGreetings,
Hawk Energy is pleased to share with you its latest energy news from NewBase Energy
as per attached file NewBase 05 May 2025 Energy News issue - 1785 by Khaled Al Awadi
Regards.
Founder & Senior Editor NewBase Energy
Khaled M Al Awadi, Energy ConsultantGreetings,
Hawk Energy is pleased to share with you its latest energy news from NewBase Energy
as per attached file NewBase 05 May 2025 Energy News issue - 1785 by Khaled Al Awadi
Regards.
Founder & Senior Editor NewBase Energy
Khaled M Al Awadi, Energy Consultant
The Institute for Public Relations Behavioral Insights Research Center and Leger partnered on this 5th edition of the Disinformation in Society Report. We surveyed 2,000 U.S. adults to assess what sources they trust, how Americans perceive false or misleading information, who they hold responsible for spreading it, and what actions they believe are necessary to combat it.
Olga Baranets: AI Doesn’t Wait for Retros (UA)
UA Online PMDay 2025 Spring
Website – https://pmday.org/online
Youtube – https://www.youtube.com/startuplviv
FB – https://www.facebook.com/pmdayconference
Kiran Flemish is a dynamic musician, composer, and student leader pursuing a degree in music with a minor in film and media studies. As a talented tenor saxophonist and DJ, he blends jazz with modern digital production, creating original compositions using platforms like Logic Pro and Ableton Live. With nearly a decade of experience as a private instructor and youth music coach, Kiran is passionate about mentoring the next generation of musicians. He has hosted workshops, raised funds for causes like the Save the Music Foundation and Type I Diabetes research, and is eager to expand his career in music licensing and production.
Alan Stalcup is the visionary leader and CEO of GVA Real Estate Investments. In 2015, Alan spearheaded the transformation of GVA into a dynamic real estate powerhouse. With a relentless commitment to community and investor value, he has grown the company from a modest 312 units to an impressive portfolio of over 29,500 units across nine states. He graduated from Washington University in St. Louis and has honed his knowledge and know-how for over 20 years.
www.visualmedia.com digital markiting (1).pptxDavinder Singh
Visual media is a visual way of communicating meaning. This includes digital media such as social media and traditional media such as television. Visual media can encompass entertainment, advertising, art, performance art, crafts, information artifacts and messages between people.
Smart Home Market Size, Growth and Report (2025-2034)GeorgeButtler
The global smart home market was valued at approximately USD 52.01 billion in 2024. Driven by rising consumer demand for automation, energy efficiency, and enhanced security, the market is expected to expand at a CAGR of 15.00% from 2025 to 2034. By the end of the forecast period, it is projected to reach around USD 210.41 billion, reflecting significant growth opportunities across emerging and developed regions as smart technologies continue to transform residential living environments.
**Title:** Accounting Basics – A Complete Visual Guide
**Author:** CA Suvidha Chaplot
**Description:**
Whether you're a beginner in business, a commerce student, or preparing for professional exams, understanding the language of business — **accounting** — is essential. This beautifully designed SlideShare simplifies key accounting concepts through **colorful infographics**, clear examples, and smart layouts.
From understanding **why accounting matters** to mastering **core principles, standards, types of accounts, and the accounting equation**, this guide covers everything in a visual-first format.
📘 **What’s Inside:**
* **Introduction to Accounting**: Definition, objectives, scope, and users
* **Accounting Concepts & Principles**: Business Entity, Accruals, Matching, Going Concern, and more
* **Types of Accounts**: Asset, Liability, Equity explained visually
* **The Accounting Equation**: Assets = Liabilities + Equity broken down with diagrams
* BONUS: Professionally designed cover for presentation or academic use
🎯 **Perfect for:**
* Students (Commerce, BBA, MBA, CA Foundation)
* Educators and Trainers
* UGC NET/Assistant Professor Aspirants
* Anyone building a strong foundation in accounting
👩🏫 **Designed & curated by:** CA Suvidha Chaplot
Discover how to use web scraping for social media activity tracking. Let’s discuss steps, tools and techniques to track social media activities –competitor activities, brand mentions, consumer preferences, and trends.
This blog explores the impactful leadership of Kunal Bansal, Director of GMI Infra, highlighting his role in community development through events like the KOMPTE Badminton Tournament 2025 in Chandigarh. A reflection on how infrastructure and social responsibility go hand-in-hand in building the future.
Brandon Flatley masterfully blends creativity and community impact. As a mixologist and small business owner, he delivers unforgettable cocktail experiences. A musician at heart, he excels in composition and recording.
5. family tree openpbx telephony free and open source proprietary asterisk gnu bayonne yate astlinux trixbox xorcom rapid xorcom rapid elastix VoIPonCD
7. acronyms VoIP – Voice over Internet Protocol POTS – Plain Old Telephone Service ATA – Analog Telephone Adapter WiFi – Wireless Fidelity SIP – Session Initiation Protocol IAX – Inter-Asterisk eXchange PBX – Private Branch eXchange SBC – Single Board Computer
8. advantages Network based – digital by design IP based – economies of scale and scope Crossover to e-mail, IM, etc. CRM integration – e.g. SugarCRM No toll boundaries – IP goes everywhere One infrastructure to worry about
9. disadvantages New – adoption issues Migration from legacy systems will cost $$$ Innovative – requires a big change How will telcos overbill? ???
10. design as a network application If you assess your requirements from a network perspective, the design is more intuitive Think of VoIP as an application along the lines of e-mail Accounts follow a <name>@<registrar> syntax Account may map to a 7 or 10 digit phone number if system relays to POTS lines
11. asterisk – the project Asterisk was originally written by Mark Spencer of Digium, Inc. Released under GPL. Project began in 1999 or so. Core PBX + support services such as voicemail, call-forwarding, conference calling, etc.
13. scenario 1 Hobby/home use No POTS lines involved. Extensions for family and friends. All calls happen on PCs or ATAs.
14. scenario 2 Small business One or two POTS lines for incoming/outgoing calls. Extensions for Tech Support, Sales, Customer Service. Generic operator extension. Numeric extensions and voice mail for employees.
15. scenario 3 Small to mid-size business Multiple POTS and VoIP lines for incoming/outgoing calls. Aggregation of lines for multiple calls (rollover dialing). Specific lines and providers for long-distance and international calling.
16. astlinux Linux 2.6 Soekris and WRAP Asterisk 1.2 mini_httpd + PHP in CGI mode OpenSSH OpenSSL "keydisk" support traffic shaper (iptables+tc) tftp server "PBX Only Mode" - Asterisk only ftp server (vsftpd) Sangoma A101/102/104/S518 support SNMPD for lm_sensors rp-pppoe
17. astlinux Embedded Linux for Asterisk Live CD ISO is about 70 MB CF card fits under 64 MB Targeted for Soekris or WRAP SBCs Needs Pentium class 266 MHz or better GUI via web browser All scripts are editable via textarea boxes on web pages. Additional stats, PHP, httpd, available via browser
18. open source maturity model OSMM is a trademark of Navica . A chart like this will make you a hero at work. PHB image used without permission
29. more complex solutions Trixbox Used to be Asterisk @ Home Combines several open source projects into one distro. CentOS based Caution: Wipes out the entire hard drive!!! If you simply want to “play” with VoIP, use AstLinux live CD.
30. dev kit used in demo TDM400P from Digium Photo from http://www.marlow.dk/images/asterisk/tdm400p-fxo-small.jpg IAXy S101 from Digium Photo from http://www.voip-info.org/users/385/25385/images/618/IAXy.jpg