The document discusses Voice over IP (VoIP) and IP-PBXs. It describes how IP-PBXs integrate voice and data communication over a single IP network, replacing separate phone and computer networks. This consolidation provides cost savings through reduced infrastructure needs and more efficient use of bandwidth. However, VoIP faces challenges in ensuring call quality and reliability. The document also outlines several applications of IP-PBXs, such as unified messaging systems, software-based IP phones, and enhanced call routing using computer calendar and login data.
A NEW SYSTEM ON CHIP RECONFIGURABLE GATEWAY ARCHITECTURE FOR VOICE OVER INTER...csandit
The aim of this paper is to present a new System on Chip (SoC) reconfigurable gateway
architecture for Voice over Internet Telephony (VOIP). Our motivation behind this work is
justified by the following arguments: most of VOIP solutions proposed in the market are based
on the use of a general purpose processor and a DSP circuit. In these solutions, the use of the
serial multiply accumulate circuit is very limiting for the signal processing. Also, in embedded
VOIP based DSP applications, the DSP works without MMU (memory management unit). This
is a serious limitation because VOIP solutions are multi-task based. In order to overcome these
problems, we propose a new VOIP gateway architecture built around the OpenRisc-1200-V3
processor. This last one integrates a DSP circuit as well as a MMU. The hardware architecture
is mapped into the VIRTEX-5 FPGA device. We propose a design methodology based on the
design for reuse and design with reuse concepts. We demonstrate that the proposed SoC
architecture is reconfigurable, scalable and the final RTL code can be reused for any FPGA or
ASIC technology. Performances measures, in the VIRTEX-5 FPGA device family, show that the
SOC-gateway architecture occupies 52% of the FPGA in term of slice LUT, 42% of IOBs, 60%
of bloc memory, 8% of integrated DSP, 16% of PLL and the total power is estimated at
4.3Watts.
A NEW SYSTEM ON CHIP RECONFIGURABLE GATEWAY ARCHITECTURE FOR VOICE OVER INTER...cscpconf
The aim of this paper is to present a new System on Chip (SoC) reconfigurable gateway architecture for Voice over Internet Telephony (VOIP). Our motivation behind this work is
justified by the following arguments: most of VOIP solutions proposed in the market are based on the use of a general purpose processor and a DSP circuit. In these solutions, the use of the serial multiply accumulate circuit is very limiting for the signal processing. Also, in embedded VOIP based DSP applications, the DSP works without MMU (memory management unit). This is a serious limitation because VOIP solutions are multi-task based. In order to overcome these
problems, we propose a new VOIP gateway architecture built around the OpenRisc-1200-V3 processor. This last one integrates a DSP circuit as well as a MMU. The hardware architecture is mapped into the VIRTEX-5 FPGA device. We propose a design methodology based on the design for reuse and design with reuse concepts. We demonstrate that the proposed SoC architecture is reconfigurable, scalable and the final RTL code can be reused for any FPGA or ASIC technology. Performances measures, in the VIRTEX-5 FPGA device family, show that the SOC-gateway architecture occupies 52% of the FPGA in term of slice LUT, 42% of IOBs, 60% of bloc memory, 8% of integrated DSP, 16% of PLL and the total power is estimated at 4.3Watts
Deepak Rajpoot is a telecommunications professional with over 6 years of experience in project management, network implementation, and quality management for the installation and commissioning of telecom networks. He has worked for various companies including Reliance Jio Infocomm, Teleysia Network Solutions, L&T InfoTech, and ZTE Telecom on projects involving GSM, WCDMA, LTE, and 4G networks. Currently he works as an Assistant Manager at Reliance Jio Infocomm providing NOC TAC support.
This document describes a system that uses a Raspberry Pi to help visually impaired people read printed text. The system works by using a camera to take photos of printed text, optical character recognition software to convert the images to text files, and text-to-speech software to convert the text to audio that is played through speakers. The key components are the Raspberry Pi, camera, HDMI to VGA converter, power supply, and software for OCR and text-to-speech. The system aims to make printed text accessible for visually impaired and illiterate people in a portable, low-cost manner.
A distributed ip based telecommunication system using sipIJCNCJournal
Voice over Internet Protocol (VoIP) technologies are integral to modern telecommunications because of
their advanced features, flexibility, and economic benefits. Internet Service Providers initially promoted
these technologies by providing low cost local and international calling. At present, there is also a great
deal of interest in using IP-based technologies to replace traditional small and large office telephone
systems that use traditional PBX’s (Private Branch eXchange). Unfortunately, the large majority of the
emerging VoIP based office telephone systems have followed the centralized design of traditional public
and private telephone systems in which all the intelligence in the system is at the core, with quite expensive
hardware and software components and appropriate redundancy for adequate levels of reliability. In this
paper, it is argued that a centralized model for an IP-based telecommunications system fails to exploit the
full capabilities of Internet-inspired communications and that, very simple, inexpensive, elegant and
flexible solutions are possible by deliberately avoiding the centralized approach. This paper describes the
design, philosophy and implementation of a prototype for a fully distributed IP-based Telecommunication
System (IPTS) that provides the essential feature set for office and home telecommunications, including IPbased
long-distance and local calling, and with the support for video as well as data and text. The
prototype system was implemented with an Internet-inspired distributed design using open source software,
with appropriate customizations and configurations.
Sagar Ware has over 9 years of experience in the VOIP domain working with companies like TATA Elxsi, Avaya, and Polycom. He has extensive experience in C/C++ and networking protocols like SIP, H323, and BFCP. He has worked as a team lead and individual contributor on projects involving VOIP devices, telepresence solutions, and docking stations. His roles have included requirements gathering, documentation, estimates, implementations, testing, and customer support.
Voice over Internet Protocol with Novel Applicationsirjes
This document summarizes Voice over Internet Protocol (VoIP) technology. It discusses three types of VoIP calls: PC to PC, PC to phone, and phone to phone. It also explains some key VoIP protocols like H.323, Session Initiation Protocol (SIP), and VoiceXML. The document provides an overview of how VoIP works, its advantages over traditional phone systems, and examples of VoIP applications.
The document discusses Voice over IP (VoIP) and IP-PBXs. It describes how IP-PBXs integrate voice and data communication over a single IP network, replacing separate phone and computer networks. This consolidation provides cost savings through reduced infrastructure needs and more efficient use of bandwidth. However, VoIP faces challenges in ensuring call quality and reliability. The document also outlines several applications of IP-PBXs, such as unified messaging systems, software-based IP phones, and enhanced call routing using computer calendar and login data.
A NEW SYSTEM ON CHIP RECONFIGURABLE GATEWAY ARCHITECTURE FOR VOICE OVER INTER...csandit
The aim of this paper is to present a new System on Chip (SoC) reconfigurable gateway
architecture for Voice over Internet Telephony (VOIP). Our motivation behind this work is
justified by the following arguments: most of VOIP solutions proposed in the market are based
on the use of a general purpose processor and a DSP circuit. In these solutions, the use of the
serial multiply accumulate circuit is very limiting for the signal processing. Also, in embedded
VOIP based DSP applications, the DSP works without MMU (memory management unit). This
is a serious limitation because VOIP solutions are multi-task based. In order to overcome these
problems, we propose a new VOIP gateway architecture built around the OpenRisc-1200-V3
processor. This last one integrates a DSP circuit as well as a MMU. The hardware architecture
is mapped into the VIRTEX-5 FPGA device. We propose a design methodology based on the
design for reuse and design with reuse concepts. We demonstrate that the proposed SoC
architecture is reconfigurable, scalable and the final RTL code can be reused for any FPGA or
ASIC technology. Performances measures, in the VIRTEX-5 FPGA device family, show that the
SOC-gateway architecture occupies 52% of the FPGA in term of slice LUT, 42% of IOBs, 60%
of bloc memory, 8% of integrated DSP, 16% of PLL and the total power is estimated at
4.3Watts.
A NEW SYSTEM ON CHIP RECONFIGURABLE GATEWAY ARCHITECTURE FOR VOICE OVER INTER...cscpconf
The aim of this paper is to present a new System on Chip (SoC) reconfigurable gateway architecture for Voice over Internet Telephony (VOIP). Our motivation behind this work is
justified by the following arguments: most of VOIP solutions proposed in the market are based on the use of a general purpose processor and a DSP circuit. In these solutions, the use of the serial multiply accumulate circuit is very limiting for the signal processing. Also, in embedded VOIP based DSP applications, the DSP works without MMU (memory management unit). This is a serious limitation because VOIP solutions are multi-task based. In order to overcome these
problems, we propose a new VOIP gateway architecture built around the OpenRisc-1200-V3 processor. This last one integrates a DSP circuit as well as a MMU. The hardware architecture is mapped into the VIRTEX-5 FPGA device. We propose a design methodology based on the design for reuse and design with reuse concepts. We demonstrate that the proposed SoC architecture is reconfigurable, scalable and the final RTL code can be reused for any FPGA or ASIC technology. Performances measures, in the VIRTEX-5 FPGA device family, show that the SOC-gateway architecture occupies 52% of the FPGA in term of slice LUT, 42% of IOBs, 60% of bloc memory, 8% of integrated DSP, 16% of PLL and the total power is estimated at 4.3Watts
Deepak Rajpoot is a telecommunications professional with over 6 years of experience in project management, network implementation, and quality management for the installation and commissioning of telecom networks. He has worked for various companies including Reliance Jio Infocomm, Teleysia Network Solutions, L&T InfoTech, and ZTE Telecom on projects involving GSM, WCDMA, LTE, and 4G networks. Currently he works as an Assistant Manager at Reliance Jio Infocomm providing NOC TAC support.
This document describes a system that uses a Raspberry Pi to help visually impaired people read printed text. The system works by using a camera to take photos of printed text, optical character recognition software to convert the images to text files, and text-to-speech software to convert the text to audio that is played through speakers. The key components are the Raspberry Pi, camera, HDMI to VGA converter, power supply, and software for OCR and text-to-speech. The system aims to make printed text accessible for visually impaired and illiterate people in a portable, low-cost manner.
A distributed ip based telecommunication system using sipIJCNCJournal
Voice over Internet Protocol (VoIP) technologies are integral to modern telecommunications because of
their advanced features, flexibility, and economic benefits. Internet Service Providers initially promoted
these technologies by providing low cost local and international calling. At present, there is also a great
deal of interest in using IP-based technologies to replace traditional small and large office telephone
systems that use traditional PBX’s (Private Branch eXchange). Unfortunately, the large majority of the
emerging VoIP based office telephone systems have followed the centralized design of traditional public
and private telephone systems in which all the intelligence in the system is at the core, with quite expensive
hardware and software components and appropriate redundancy for adequate levels of reliability. In this
paper, it is argued that a centralized model for an IP-based telecommunications system fails to exploit the
full capabilities of Internet-inspired communications and that, very simple, inexpensive, elegant and
flexible solutions are possible by deliberately avoiding the centralized approach. This paper describes the
design, philosophy and implementation of a prototype for a fully distributed IP-based Telecommunication
System (IPTS) that provides the essential feature set for office and home telecommunications, including IPbased
long-distance and local calling, and with the support for video as well as data and text. The
prototype system was implemented with an Internet-inspired distributed design using open source software,
with appropriate customizations and configurations.
Sagar Ware has over 9 years of experience in the VOIP domain working with companies like TATA Elxsi, Avaya, and Polycom. He has extensive experience in C/C++ and networking protocols like SIP, H323, and BFCP. He has worked as a team lead and individual contributor on projects involving VOIP devices, telepresence solutions, and docking stations. His roles have included requirements gathering, documentation, estimates, implementations, testing, and customer support.
Voice over Internet Protocol with Novel Applicationsirjes
This document summarizes Voice over Internet Protocol (VoIP) technology. It discusses three types of VoIP calls: PC to PC, PC to phone, and phone to phone. It also explains some key VoIP protocols like H.323, Session Initiation Protocol (SIP), and VoiceXML. The document provides an overview of how VoIP works, its advantages over traditional phone systems, and examples of VoIP applications.
SIP is a signaling protocol that is enabling unified communications applications by allowing IP-based communications to be extended across different networks, devices, and transport mediums. Key advantages of SIP include supporting mobility, virtual numbers, and business continuity. SIP trunking provides cost benefits like reduced network service costs by converging voice and data on a single IP connection. Considerations for SIP implementation include ensuring interoperability between systems, implementing security measures like encryption, and properly provisioning E911 services for mobile users.
The document provides an overview of a presentation on Voice over IP (VoIP) given by Tony Hutchinson. The presentation covers:
- Tony Hutchinson's background and expertise in VoIP, PBX design, and networking.
- An agenda that includes an executive summary, history of VoIP, business case for VoIP, services, convergence, and challenges.
- A high-level history of VoIP and lessons learned from previous telephony systems.
Voice over Internet Protocol (VoIP) is replacing legacy telephone networks by carrying digitized voice in IP data packets over data networks. This chapter introduces VoIP, comparing it to legacy telephone networks, and discusses VoIP standards and protocols. It also introduces WiMAX networks and discusses supporting QoS for multimedia like VoIP over WiMAX. The objectives are to guarantee QoS for multiple service classes over WiMAX and improve VoIP performance. Simulation using OPNET Modeler will analyze VoIP traffic and QoS parameters over WiMAX.
This research work investigates and improves the performance of Voice over Internet Protocol (VoIP) traffic using IPV4 and IPV6 over WiMAX networks and the impact of various voice codec schemes and statistical distribution for Voice over Internet Protocol (VoIP) over WiMAX has been investigated in detail.
Portable voice communication system on raspberry piIRJET Journal
This document summarizes a research paper on developing a portable voice communication system using a Raspberry Pi. The system uses Asterisk software to establish communication between mobile devices over WiFi. Asterisk transforms a computer into a communications server by routing voice over internet protocol (VoIP) packets. The researchers propose setting up a Raspberry Pi and laptop on a local network to test a SIP client application. Once configured, the system would allow users to make voice or video calls between phones and laptops without a SIM card or internet by assigning IP addresses and proxies. The portable system provides benefits like low cost, remote access, and reduced wiring compared to traditional PBX phone systems.
This document is a curriculum vitae for Lei Gao. It summarizes his personal and employment details. He has over 15 years of experience in the telecommunications industry, specializing in mobile networks from 2G to 4G technologies. Some of his achievements include innovating dual carrier TD-LTE solutions and demonstrating peak throughput rates for CAT3/4 devices on live networks in China. He has expertise in network optimization, performance analysis, and troubleshooting across various network elements and technologies.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
This document provides an overview of an IoT-based wireless electronic notice board project. The system will allow users to send notices from an Android phone to be displayed on an LCD screen notice board. Key aspects include:
- The main controlling device is a Raspberry Pi processor connected to a WiFi modem and LCD display. Notices sent from an Android app will be received by the WiFi modem and processed/displayed by the Raspberry Pi.
- The objectives are to develop a wireless notice board that can display notices sent from a mobile phone in a timely manner, as well as support sending documents, images, and videos.
- The existing manual notice boards have limitations like being time-consuming to maintain
Design and Simulation of VoIP System for Campus usage A Case Study at PTUijtsrd
1. The document describes the design and simulation of a VoIP system for campus usage at Pyay Technological University using Cisco Packet Tracer.
2. The network design includes routers, wireless routers, switches, and IP phones connected across 6 university departments. RIP routing protocol and VoIP technology are configured to enable voice communication between the departments.
3. The network configuration includes setting up routing, wireless access points, IP phones, and testing the voice and data connectivity across the campus network.
Looking for Mobile Application development & Software Development in India. High end training in Mobile Application development , Embedded Systems , Java , C/C++ , Microsoft.Net & SQT
This document summarizes a presentation on 5G mobile technology given by two students. It includes an abstract, introduction, sections on 2G-5G networks and 5G network architecture. The key points are:
- 5G will allow unprecedented call volume and data transmission speeds over VOIP-enabled devices.
- 5G network architecture uses a "Nano core" concept with flat IP, nanotechnology, cloud computing and all IP platforms to allow different radio access networks to upgrade to a single core network in a flexible way.
- The introduction discusses the evolution from 1G to 5G and increasing phone capabilities, with 5G allowing broadband internet access over mobile phones.
This document summarizes a presentation on Voice over Internet Protocol (VoIP) technology. It discusses what VoIP is, the advantages it provides over traditional phone systems, and ways that VoIP can be implemented. It also addresses some challenges to adopting VoIP in India, such as insufficient bandwidth in some areas and lack of policies encouraging investment. The overall purpose is to explore how VoIP can offer a more cost-effective alternative to traditional phone networks.
Voice over Internet Protocol (Voice over IP, VoIP and IP telephony) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. The terms Internet telephony, broadband telephony, and broadband phone service specifically refer to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).The steps and principals involved in originating VoIP telephone calls are similar to traditional digital telephony and involve signaling, channel setup, digitization of the analog voice signals, and encoding.
Adding P2P support to WengoPhone, an open-source VoIP and ...Videoguy
This document proposes adding peer-to-peer (P2P) support to the open-source VoIP and video conferencing application WengoPhone. The author plans to implement a P2P-SIP architecture using a Chord distributed hash table within a hierarchical network of super nodes. Key aspects to be addressed include integrating the P2P layer into the SIP protocol stack, supporting locality and proximity, and evaluating security through analyzing potential attacks and implementing countermeasures. The proposed timeline is to complete the work by the end of April 2006, with evaluation based on scalability testing and damage assessment from conducted attack scenarios.
This document discusses implementing a voting system through IP telephony. Key points include:
1. IP telephony equipment like phones, servers and gateways would be used to set up the system and allow voting through phone keypads or dedicated voting pads.
2. A voting server with a database would collect and count votes, ranking candidates based on vote totals.
3. Challenges include applying various voice and data network standards to a converged network and ensuring the same reliability as traditional voice networks.
4. Providing voting services through IP telephony could be profitable for service providers and beneficial for users through reduced costs.
This document discusses a voice-controlled robot system with two embedded systems - a master system to control the robot and a slave robot system that responds to voice commands from the master. The systems communicate wirelessly using a protocol like Bluetooth. The slave robot uses Sphinx-4 speech recognition software to understand voice commands. Video, audio and other data can be transmitted between the systems. The paper focuses on the wireless transmission of data between the embedded systems and enhancing voice processing technologies for robot control.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
This document discusses Voice over Internet Protocol (VoIP) and its use in mobile communication networks. It provides details on VOIP functionality, implementation, reliability, quality of service, difficulties with faxing, integration into the global telephone numbering system, use on mobile phones and handheld devices, security considerations, and adoption of VOIP technology. The document examines the benefits of using VOIP in mobile networks, including IP backbone networks, redundancy, and technical requirements for supporting IP traffic. It also outlines VoIP architecture and provides references.
This document summarizes a research paper on VoIP security. It discusses two important VoIP protocols, SIP and H.323, and analyzes their security features. It also defines new functionality for negotiating security mechanisms between SIP entities. Several security issues are identified for VoIP, such as denial of service attacks. Solutions to improve VoIP security include encryption at endpoints, using SRTP, better scheduling, and resolving NAT/IPsec incompatibilities.
This document is a project report on VoIP technology from Gollis University. It discusses what VoIP is, the history and evolution of VoIP, how VoIP works by breaking voice signals into packets and sending them over IP networks, common VoIP protocols and codecs, benefits of VoIP compared to traditional phone systems, and potential future enhancements to VoIP technology. It was created by a group of 6 students for their 7th semester mini project.
Audio documents can be summarized in 3 sentences or less as follows:
The document discusses various audio coding models, techniques, and standards including entropy coding, differential coding, LPC and parametric coding, sub-band coding, and audio compression standards from ITU, ISO, and MPEG. It also covers topics like audio data rates and file sizes, psychoacoustic models, compression algorithm requirements, and structured audio coding techniques like MIDI. The models, techniques, and standards discussed aim to compress audio signals while maintaining sufficient quality for the intended application.
The document summarizes key concepts about compiler design. It defines a compiler as a program that translates source code written in one language into an equivalent executable program in another language. Compilers are classified based on the number of passes, such as single-pass and multi-pass compilers. The document also discusses the analysis-synthesis model of compilation, which involves analyzing the source program to create an intermediate representation, then synthesizing target code from that representation. Major phases of a compiler include lexical analysis, parsing, semantic analysis, code generation, and optimization.
SIP is a signaling protocol that is enabling unified communications applications by allowing IP-based communications to be extended across different networks, devices, and transport mediums. Key advantages of SIP include supporting mobility, virtual numbers, and business continuity. SIP trunking provides cost benefits like reduced network service costs by converging voice and data on a single IP connection. Considerations for SIP implementation include ensuring interoperability between systems, implementing security measures like encryption, and properly provisioning E911 services for mobile users.
The document provides an overview of a presentation on Voice over IP (VoIP) given by Tony Hutchinson. The presentation covers:
- Tony Hutchinson's background and expertise in VoIP, PBX design, and networking.
- An agenda that includes an executive summary, history of VoIP, business case for VoIP, services, convergence, and challenges.
- A high-level history of VoIP and lessons learned from previous telephony systems.
Voice over Internet Protocol (VoIP) is replacing legacy telephone networks by carrying digitized voice in IP data packets over data networks. This chapter introduces VoIP, comparing it to legacy telephone networks, and discusses VoIP standards and protocols. It also introduces WiMAX networks and discusses supporting QoS for multimedia like VoIP over WiMAX. The objectives are to guarantee QoS for multiple service classes over WiMAX and improve VoIP performance. Simulation using OPNET Modeler will analyze VoIP traffic and QoS parameters over WiMAX.
This research work investigates and improves the performance of Voice over Internet Protocol (VoIP) traffic using IPV4 and IPV6 over WiMAX networks and the impact of various voice codec schemes and statistical distribution for Voice over Internet Protocol (VoIP) over WiMAX has been investigated in detail.
Portable voice communication system on raspberry piIRJET Journal
This document summarizes a research paper on developing a portable voice communication system using a Raspberry Pi. The system uses Asterisk software to establish communication between mobile devices over WiFi. Asterisk transforms a computer into a communications server by routing voice over internet protocol (VoIP) packets. The researchers propose setting up a Raspberry Pi and laptop on a local network to test a SIP client application. Once configured, the system would allow users to make voice or video calls between phones and laptops without a SIM card or internet by assigning IP addresses and proxies. The portable system provides benefits like low cost, remote access, and reduced wiring compared to traditional PBX phone systems.
This document is a curriculum vitae for Lei Gao. It summarizes his personal and employment details. He has over 15 years of experience in the telecommunications industry, specializing in mobile networks from 2G to 4G technologies. Some of his achievements include innovating dual carrier TD-LTE solutions and demonstrating peak throughput rates for CAT3/4 devices on live networks in China. He has expertise in network optimization, performance analysis, and troubleshooting across various network elements and technologies.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
This document provides an overview of an IoT-based wireless electronic notice board project. The system will allow users to send notices from an Android phone to be displayed on an LCD screen notice board. Key aspects include:
- The main controlling device is a Raspberry Pi processor connected to a WiFi modem and LCD display. Notices sent from an Android app will be received by the WiFi modem and processed/displayed by the Raspberry Pi.
- The objectives are to develop a wireless notice board that can display notices sent from a mobile phone in a timely manner, as well as support sending documents, images, and videos.
- The existing manual notice boards have limitations like being time-consuming to maintain
Design and Simulation of VoIP System for Campus usage A Case Study at PTUijtsrd
1. The document describes the design and simulation of a VoIP system for campus usage at Pyay Technological University using Cisco Packet Tracer.
2. The network design includes routers, wireless routers, switches, and IP phones connected across 6 university departments. RIP routing protocol and VoIP technology are configured to enable voice communication between the departments.
3. The network configuration includes setting up routing, wireless access points, IP phones, and testing the voice and data connectivity across the campus network.
Looking for Mobile Application development & Software Development in India. High end training in Mobile Application development , Embedded Systems , Java , C/C++ , Microsoft.Net & SQT
This document summarizes a presentation on 5G mobile technology given by two students. It includes an abstract, introduction, sections on 2G-5G networks and 5G network architecture. The key points are:
- 5G will allow unprecedented call volume and data transmission speeds over VOIP-enabled devices.
- 5G network architecture uses a "Nano core" concept with flat IP, nanotechnology, cloud computing and all IP platforms to allow different radio access networks to upgrade to a single core network in a flexible way.
- The introduction discusses the evolution from 1G to 5G and increasing phone capabilities, with 5G allowing broadband internet access over mobile phones.
This document summarizes a presentation on Voice over Internet Protocol (VoIP) technology. It discusses what VoIP is, the advantages it provides over traditional phone systems, and ways that VoIP can be implemented. It also addresses some challenges to adopting VoIP in India, such as insufficient bandwidth in some areas and lack of policies encouraging investment. The overall purpose is to explore how VoIP can offer a more cost-effective alternative to traditional phone networks.
Voice over Internet Protocol (Voice over IP, VoIP and IP telephony) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. The terms Internet telephony, broadband telephony, and broadband phone service specifically refer to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).The steps and principals involved in originating VoIP telephone calls are similar to traditional digital telephony and involve signaling, channel setup, digitization of the analog voice signals, and encoding.
Adding P2P support to WengoPhone, an open-source VoIP and ...Videoguy
This document proposes adding peer-to-peer (P2P) support to the open-source VoIP and video conferencing application WengoPhone. The author plans to implement a P2P-SIP architecture using a Chord distributed hash table within a hierarchical network of super nodes. Key aspects to be addressed include integrating the P2P layer into the SIP protocol stack, supporting locality and proximity, and evaluating security through analyzing potential attacks and implementing countermeasures. The proposed timeline is to complete the work by the end of April 2006, with evaluation based on scalability testing and damage assessment from conducted attack scenarios.
This document discusses implementing a voting system through IP telephony. Key points include:
1. IP telephony equipment like phones, servers and gateways would be used to set up the system and allow voting through phone keypads or dedicated voting pads.
2. A voting server with a database would collect and count votes, ranking candidates based on vote totals.
3. Challenges include applying various voice and data network standards to a converged network and ensuring the same reliability as traditional voice networks.
4. Providing voting services through IP telephony could be profitable for service providers and beneficial for users through reduced costs.
This document discusses a voice-controlled robot system with two embedded systems - a master system to control the robot and a slave robot system that responds to voice commands from the master. The systems communicate wirelessly using a protocol like Bluetooth. The slave robot uses Sphinx-4 speech recognition software to understand voice commands. Video, audio and other data can be transmitted between the systems. The paper focuses on the wireless transmission of data between the embedded systems and enhancing voice processing technologies for robot control.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
This document discusses Voice over Internet Protocol (VoIP) and its use in mobile communication networks. It provides details on VOIP functionality, implementation, reliability, quality of service, difficulties with faxing, integration into the global telephone numbering system, use on mobile phones and handheld devices, security considerations, and adoption of VOIP technology. The document examines the benefits of using VOIP in mobile networks, including IP backbone networks, redundancy, and technical requirements for supporting IP traffic. It also outlines VoIP architecture and provides references.
This document summarizes a research paper on VoIP security. It discusses two important VoIP protocols, SIP and H.323, and analyzes their security features. It also defines new functionality for negotiating security mechanisms between SIP entities. Several security issues are identified for VoIP, such as denial of service attacks. Solutions to improve VoIP security include encryption at endpoints, using SRTP, better scheduling, and resolving NAT/IPsec incompatibilities.
This document is a project report on VoIP technology from Gollis University. It discusses what VoIP is, the history and evolution of VoIP, how VoIP works by breaking voice signals into packets and sending them over IP networks, common VoIP protocols and codecs, benefits of VoIP compared to traditional phone systems, and potential future enhancements to VoIP technology. It was created by a group of 6 students for their 7th semester mini project.
Audio documents can be summarized in 3 sentences or less as follows:
The document discusses various audio coding models, techniques, and standards including entropy coding, differential coding, LPC and parametric coding, sub-band coding, and audio compression standards from ITU, ISO, and MPEG. It also covers topics like audio data rates and file sizes, psychoacoustic models, compression algorithm requirements, and structured audio coding techniques like MIDI. The models, techniques, and standards discussed aim to compress audio signals while maintaining sufficient quality for the intended application.
The document summarizes key concepts about compiler design. It defines a compiler as a program that translates source code written in one language into an equivalent executable program in another language. Compilers are classified based on the number of passes, such as single-pass and multi-pass compilers. The document also discusses the analysis-synthesis model of compilation, which involves analyzing the source program to create an intermediate representation, then synthesizing target code from that representation. Major phases of a compiler include lexical analysis, parsing, semantic analysis, code generation, and optimization.
The project budget document outlines the estimated costs and time-phased budget for the project. It lists one computer costing 20,000 birr as the main required material. The time-phased budget shows 5 main tasks including collecting data, preparing materials, design and programming, testing, and implementation, with estimated durations ranging from 1 to 5 days depending on optimistic, most likely, or pessimistic scenarios.
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Cs project proposal
1. Computer Science Project Proposal
1. Principal Investigator
Brendan Marlborough. Currently doing joint Computer Science and Information
Systems Honours at Rhodes University.
Address: 6 Leicester Street Grahamstown 6139.
Email: g02m1213@campus.ru.ac.za
2. Title of Project
SIP / H.323 / AIX telephone customization
3. Mentoring
Professor Peter Clayton, Research Director for Computer Science, Rhodes University
P.Clayton@ru.ac.za
Professor Alfredo Terzoli, Centre of Excellence Co-ordinator, Computer Science,
Rhodes University
A.Terzoli@ru.ac.za
4. Objective of Research
The main aim of this project is to enhance the features of the current IP telephone
used in the department. This will involve fixing certain features of the phone which
are not suitable for our purposes such as improving the display. Encryption may also
be added to the phone in conjunction with a master’s student. Another feature the
phone lacks is the ability to connect over a PPPoE network and this may also be
rectified. The following needs to be researched in order to accomplish these
objectives: What changes will have to be made to the interface (keypad and display)
2. in order to make the phone more user friendly, should encryption be added what
changes if any will have to be made to the existing network and will the phone be able
to support encryption.
5. Background/History of the Study
The Computer Science department at Rhodes University currently uses a telephone
service known as iLanga. iLanga uses Asterisk, an open source component, which is
used to connect different voice networks, including networks based on SIP and H.232
(both VoIP protocols) and also the PSTN (public switched telephone network). The
telephone being upgraded is one type of end device used in iLanga. The phone is
manufactured and programmed in China and comes with an API which can be used to
re-program it. These telephones will soon be offered to a limited number of students
in residences at Rhodes University. For this reason the phone should be as user
friendly as possible. It would also be preferable that the phone has encryption if it is
to be used for private conversations, and if the phone is to be used via a wireless
network then it would need to support PPPoE.
6. Approach to the Study
Reading will have to be done on how the voice network in the Computer Science
department works, how the phone connects and makes calls via the VoIP gateways
and also how the protocols the phone uses work. Reading will also have to be done
about embedded systems and how to update the flash memory in the phone. Should
encryption be added reading will be have to be done on that subject.
7. Information to be Derived/Deliverables
The project aims to improve some features of the phone which are lacking. The phone
is a very cheap option for our purposes despite some of its bad and lacking features.
Ways to improve the phones interface should therefore be found and implemented in
order to make the phone more suitable for our use and the use in residences.
3. 8. Equipment Requirements
A VoIP telephone with all the tools that will be used to compile the new C programs
and update the phones flash memory. A connection to the Rhodes voice network in
order to test the phone is also required.
9. Initial timeline for implementation
1 Week Learn how the various VoIP protocols work.
3 Weeks Learn more about embedded programming and
understand the code provided with the phone.
1 Week Find ways to improve the interface.
3 Weeks Implement changes to the interface.
1 Week Investigate the viability of adding encryption to
the phone
2 Weeks Possibly add encryption to the phone
3 Weeks Investigate adding PPPoE and possibly
implementing it.
This is just an initial time line and not enough detail is known to provide a better time
estimate. These steps may not be carried out in the same order as they appear above
either and a more iterative approach is likely to be used.
10. References
Centrality Communications, PA1688 IP Phone Development guide, Centrality
Communications, 2004
Penton, J., Terzoli, A., Asterisk: A Converged TDM and Packet-based
Communications System, Computer Science Department, Rhodes University, 2003
Penton, J., Terzoli, A., iLanga: A Next Generation VoIP-based, TDM-enabled PBX,
South African Telecommunications Networks and Appliances Conference, September
2004, Spiers