The document provides an overview of VoIP components, standards, architectures and implementation choices. It discusses key VoIP elements like terminals, packetized voice, soft switches, media servers, gateways, LANs/WANs and standards. It also describes common VoIP architectures for computer-computer, computer-phone, phone-phone communication over the internet. Finally, it outlines VoIP solutions for businesses using VoIP-enabled PBXs, replacing PBXs with softswitches, and hosted PBX solutions.
The document proposes an architecture for establishing a distributed IP-PBX communication system using multiple voice registers on different platforms and integrating both packet-switched and circuit-switched networks. It provides background on telecommunication technologies and protocols as well as an example case study of implementing the proposed architecture for a nationwide organization with distributed regional offices connected over an IP network. The case study demonstrates configuration of an Asterisk server and Cisco routers to enable voice communication between the regional branches using both the IP network and public switched telephone network.
This seminar presentation provides an overview of Voice over Internet Protocol (VoIP) technology. It discusses how VoIP works by converting voice signals to digital signals sent over the Internet via packet switching. It covers major components of VoIP networks like codecs, quality of service issues, and types of VoIP services. The presentation also highlights advantages of VoIP like reduced costs, and discusses future directions such as increased reliability and integration with other applications. In conclusion, it predicts growing adoption of VoIP technology for computer-based communications and cost-effective multimedia transfers.
This document provides an overview of key concepts related to Voice over IP (VoIP) technology. It defines common VoIP terms and standards, describes how VoIP works by breaking analog voice signals into digital packets, and outlines typical system elements like softswitches, terminals, and gateways. It also discusses media standards, signaling protocols, quality of service measures, fax transmission methods, and various Patton Electronics VoIP products.
VoIP allows users to make voice calls over the internet instead of traditional phone lines. It works by converting voice signals to digital data packets that are transmitted over the internet and then reconverted at their destination. Key components include gateways, codecs, servers, and protocols like SIP and H.323. VoIP offers advantages like lower costs and integration with other systems but relies on internet connectivity and faces some security risks.
This document describes four business models for VOIP services. Model 1 involves a VOIP provider offering PC-to-phone and PC-to-PC services, with a distributor providing VOIP codes and cards. Model 2 is similar but has the distributor handling PC-to-PC calls. Model 3 splits functions between the provider and distributor. Model 4 uses multiple providers offering different international call rates, with the distributor selecting the best provider.
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.
Kwader is a technical business organization in Saudi Arabia that specializes in providing IT solutions including IP-PBXs. An IP-PBX is a phone system that allows a business to share phone lines across locations using an IP network rather than separate phone lines. IP-PBXs provide advantages like cost savings, easy management, mobility features, and integration of technologies like video conferencing. Grandstream is a provider of IP-PBX and VoIP solutions that Kwader represents to help businesses improve communication and productivity.
VoIP allows for transmitting voice calls over TCP/IP networks instead of traditional circuit-switched networks. It started gaining popularity in the mid-1990s but had drawbacks due to lack of broadband. VoIP offers unlimited distance, lower costs, and uses standards-based protocols like H.323, SIP, and MGCP. Tadiran deployed VoIP across multiple sites globally using Universal Gateways and IP phones.
The document discusses 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.
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.
This document provides an overview and update on AudioCodes session border controller (SBC) products. It summarizes that AudioCodes is a market leader in SBCs, with the fastest growing market share. It highlights key SBC products like the Mediant 9000 and features such as advanced routing management, global partner strategy, and a comprehensive product portfolio. The document aims to showcase AudioCodes' SBC technology and momentum in the enterprise voice and data market.
Practical Fundamentals of Voice over IP (VoIP) for Engineers and TechniciansLiving Online
油
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
This document discusses Voice over Internet Protocol (VoIP). It begins by introducing VoIP and how it allows phone calls and faxes to be sent over IP-based data networks. It then discusses how VoIP works by digitizing voice, compressing it into packets, transmitting the packets over the internet, and reconstructing the voice signal at the receiving end. The document also covers some key components of a VoIP system such as encoders, decoders, and quality of service mechanisms. Finally, it briefly mentions that most VoIP implementations follow the ITU H.323 standard.
Fibernetics offers a PBX phone system called the Fibernetics Digital PBX that allows businesses to eliminate monthly phone line charges. As a competitive local exchange carrier, Fibernetics operates its own private voice and data networks that are directly connected to the public switched telephone network. The Fibernetics Digital PBX utilizes this network to provide a full-featured phone system with toll-quality voice and high reliability over internet protocol connections while requiring less bandwidth than typical VoIP systems.
The document discusses the Session Initiation Protocol (SIP), which allows for multimedia communication sessions over IP networks. SIP establishes sessions for voice, video, messaging and other applications. It uses requests and responses to initiate sessions between users, locate users, invite them to sessions, and terminate sessions. SIP relies on user agents, proxy servers, redirect servers and registrar servers. It enables mobility and flexibility in setting up and modifying communication sessions across different devices.
Voice over Internet Protocol (VoIP) allows users to make voice calls using an Internet connection instead of a regular phone line. VoIP converts voice signals from phones into digital data packets that can be transmitted over the Internet or a private IP network. Various companies offer VoIP phone service with features like voicemail, caller ID, call forwarding and more that work similar to traditional phone service but utilize an Internet connection.
Voice over Internet Protocol (VoIP) allows users to make voice calls using an Internet connection instead of a regular phone line. VoIP converts voice signals from phones into digital data packets that can be transmitted over the Internet or a private IP network. Major VoIP providers have adopted various business models and technical solutions for transmitting calls, including the use of codecs to compress audio and video, and features like voicemail, caller ID, and call forwarding.
This document summarizes information presented by AudioCodes about migrating to SIP trunking. It discusses:
1) How AudioCodes solutions like media gateways and session border controllers enable businesses to migrate phone systems to SIP trunking gradually, starting with a trial that reduces reliance on PSTN lines.
2) The cost savings that SIP trunking can provide compared to traditional phone lines, often 25-50% lower recurring charges through elimination of dual infrastructure and competitive SIP trunking markets.
3) A promotional offer from AudioCodes for VARs, which provides discounted hardware and support, as well as SIP training and certification, to help customers evaluate SIP trunking.
BriCom provides software and hardware solutions that allow different radio systems and devices to interconnect over IP networks. This includes linking radio sites anywhere in the world and enabling legacy radios to connect to the internet. Their product line includes hardware devices that connect radios to IP networks and software applications that turn smartphones and computers into virtual radios. The solutions aim to improve communications for mobile workers and first responders.
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 allows users to make phone calls using an Internet connection rather than a traditional phone line. It works by converting the voice signal from analog to digital, breaking it into packets, sending it over IP, reassembling it at the destination, and converting it back to analog. VoIP has advantages like low cost and portability but disadvantages like quality issues during power outages or network instability. Major challenges include addressing latency, echo, jitter, connection problems through firewalls and NAT, and overall reliability.
This document provides an overview of VoIP services through a seminar presentation. It discusses how VoIP came about as an alternative to traditional circuit-switched telephony using the PSTN. VoIP allows carrying voice calls over an IP network by digitizing and packetizing voice streams using protocols like SIP and H.323. Some key benefits of VoIP include reduced costs, increased flexibility, and mobility. Popular VoIP service providers include Skype, while security poses ongoing challenges to VoIP adoption.
The document summarizes an IP phone system called Allworx that is designed for businesses. It provides diverse voice services, robust features, and global integration capabilities across multiple sites. Allworx systems aim to satisfy customer needs with low total cost of ownership through easy installation and maintenance.
VOICE OVER INTERNET PROTOCOL (VOIP) allows users to make phone calls using an Internet connection rather than a traditional phone line. VOIP compresses voice data, converts it to digital signals in IP packets, and transports them over the Internet or data networks. This provides economic benefits compared to traditional phone networks since the same infrastructure can be used for both data and voice. While VOIP provides advantages like low cost and ability to communicate anywhere, it faces challenges around voice delays and compatibility with existing phone networks. Key components of a VOIP system include clients, servers, and gateways to connect VOIP and traditional phone networks.
This session will provide a quick review of the methodology of early dispatch systems connected to radio, telephone and other resources via circuit switched interfaces such as 4WE&M, 2W analogue etc., and their restricted backhaul capabilities, leading on to the 'stand-alone' RoIP boxes that allowed 4W E&M to be converted to IP and recovered at the other end allowing backhaul via more flexible IP networks.
The next technology is dispatch systems with native IP connectivity allowing the most flexible and functional interfaces between the dispatch system and its connected resources. While some manufacturers equipment uses proprietary IP messaging, most prefer and use open standards such as P25 CSSI (console sub system interface), DFSI (digital fixed system interface) and ISSI (inter sub system interface) or the emerging DMR AIS which ensures that different vendors equipment can interoperate with each other via these interfaces. Open standards provide end users with greatly improved competitive choice and functional capability on these systems.
The session will explore examples of IP interfaces for voice dispatch systems and the functions supported, plus give a background on how these apply to many different technologies and can even be adapted for conventional radio applications:
The workshop will cover on the following issues:
- The difference between RoIP and VoIP - how radio systems differ from phone systems
- Implementing one-to-one connections
- Implementing many-to-many connection
- Risk management: Identifying network issues affecting RoIP/VoIP quality; maintenance; and redundancy
- Design elements :- building blocks; calculating network bandwidth requirements
The implications of RoIP for dispatch consoles will be also be discussed: how dispatch console to radio connections can be implemented with RoIP and how RoIP can be used to provide fault tolerant dispatch architectures.
Finally the workshop will look at the impact of new technologies such as IPv6, Wireless Broadband and the switch to Digital Radio on the RoIP landscape.
Les Scott, Manager, System Sales, Zetron
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.
Voice over Internet Protocol (VoIP), is a technology that allows to make voice calls using a broadband Internet connection instead of a regular (or analog) phone line (PSTN).
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
This document provides an overview of VoIP security. It discusses the basics of VoIP security including authentication, authorization, availability, and encryption. It outlines some common attack vectors such as accessing an unsecured local network connection, wireless network, or public network. It also mentions threats from compromising a phone's configuration file or uploading a malicious file. The document summarizes some unconventional VoIP security threats like phishing, caller ID spoofing, eavesdropping, call redirection, and spam over internet telephony.
Voice over IP (VoIP) 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. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service.
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.
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.
This document provides an overview and update on AudioCodes session border controller (SBC) products. It summarizes that AudioCodes is a market leader in SBCs, with the fastest growing market share. It highlights key SBC products like the Mediant 9000 and features such as advanced routing management, global partner strategy, and a comprehensive product portfolio. The document aims to showcase AudioCodes' SBC technology and momentum in the enterprise voice and data market.
Practical Fundamentals of Voice over IP (VoIP) for Engineers and TechniciansLiving Online
油
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
This document discusses Voice over Internet Protocol (VoIP). It begins by introducing VoIP and how it allows phone calls and faxes to be sent over IP-based data networks. It then discusses how VoIP works by digitizing voice, compressing it into packets, transmitting the packets over the internet, and reconstructing the voice signal at the receiving end. The document also covers some key components of a VoIP system such as encoders, decoders, and quality of service mechanisms. Finally, it briefly mentions that most VoIP implementations follow the ITU H.323 standard.
Fibernetics offers a PBX phone system called the Fibernetics Digital PBX that allows businesses to eliminate monthly phone line charges. As a competitive local exchange carrier, Fibernetics operates its own private voice and data networks that are directly connected to the public switched telephone network. The Fibernetics Digital PBX utilizes this network to provide a full-featured phone system with toll-quality voice and high reliability over internet protocol connections while requiring less bandwidth than typical VoIP systems.
The document discusses the Session Initiation Protocol (SIP), which allows for multimedia communication sessions over IP networks. SIP establishes sessions for voice, video, messaging and other applications. It uses requests and responses to initiate sessions between users, locate users, invite them to sessions, and terminate sessions. SIP relies on user agents, proxy servers, redirect servers and registrar servers. It enables mobility and flexibility in setting up and modifying communication sessions across different devices.
Voice over Internet Protocol (VoIP) allows users to make voice calls using an Internet connection instead of a regular phone line. VoIP converts voice signals from phones into digital data packets that can be transmitted over the Internet or a private IP network. Various companies offer VoIP phone service with features like voicemail, caller ID, call forwarding and more that work similar to traditional phone service but utilize an Internet connection.
Voice over Internet Protocol (VoIP) allows users to make voice calls using an Internet connection instead of a regular phone line. VoIP converts voice signals from phones into digital data packets that can be transmitted over the Internet or a private IP network. Major VoIP providers have adopted various business models and technical solutions for transmitting calls, including the use of codecs to compress audio and video, and features like voicemail, caller ID, and call forwarding.
This document summarizes information presented by AudioCodes about migrating to SIP trunking. It discusses:
1) How AudioCodes solutions like media gateways and session border controllers enable businesses to migrate phone systems to SIP trunking gradually, starting with a trial that reduces reliance on PSTN lines.
2) The cost savings that SIP trunking can provide compared to traditional phone lines, often 25-50% lower recurring charges through elimination of dual infrastructure and competitive SIP trunking markets.
3) A promotional offer from AudioCodes for VARs, which provides discounted hardware and support, as well as SIP training and certification, to help customers evaluate SIP trunking.
BriCom provides software and hardware solutions that allow different radio systems and devices to interconnect over IP networks. This includes linking radio sites anywhere in the world and enabling legacy radios to connect to the internet. Their product line includes hardware devices that connect radios to IP networks and software applications that turn smartphones and computers into virtual radios. The solutions aim to improve communications for mobile workers and first responders.
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 allows users to make phone calls using an Internet connection rather than a traditional phone line. It works by converting the voice signal from analog to digital, breaking it into packets, sending it over IP, reassembling it at the destination, and converting it back to analog. VoIP has advantages like low cost and portability but disadvantages like quality issues during power outages or network instability. Major challenges include addressing latency, echo, jitter, connection problems through firewalls and NAT, and overall reliability.
This document provides an overview of VoIP services through a seminar presentation. It discusses how VoIP came about as an alternative to traditional circuit-switched telephony using the PSTN. VoIP allows carrying voice calls over an IP network by digitizing and packetizing voice streams using protocols like SIP and H.323. Some key benefits of VoIP include reduced costs, increased flexibility, and mobility. Popular VoIP service providers include Skype, while security poses ongoing challenges to VoIP adoption.
The document summarizes an IP phone system called Allworx that is designed for businesses. It provides diverse voice services, robust features, and global integration capabilities across multiple sites. Allworx systems aim to satisfy customer needs with low total cost of ownership through easy installation and maintenance.
VOICE OVER INTERNET PROTOCOL (VOIP) allows users to make phone calls using an Internet connection rather than a traditional phone line. VOIP compresses voice data, converts it to digital signals in IP packets, and transports them over the Internet or data networks. This provides economic benefits compared to traditional phone networks since the same infrastructure can be used for both data and voice. While VOIP provides advantages like low cost and ability to communicate anywhere, it faces challenges around voice delays and compatibility with existing phone networks. Key components of a VOIP system include clients, servers, and gateways to connect VOIP and traditional phone networks.
This session will provide a quick review of the methodology of early dispatch systems connected to radio, telephone and other resources via circuit switched interfaces such as 4WE&M, 2W analogue etc., and their restricted backhaul capabilities, leading on to the 'stand-alone' RoIP boxes that allowed 4W E&M to be converted to IP and recovered at the other end allowing backhaul via more flexible IP networks.
The next technology is dispatch systems with native IP connectivity allowing the most flexible and functional interfaces between the dispatch system and its connected resources. While some manufacturers equipment uses proprietary IP messaging, most prefer and use open standards such as P25 CSSI (console sub system interface), DFSI (digital fixed system interface) and ISSI (inter sub system interface) or the emerging DMR AIS which ensures that different vendors equipment can interoperate with each other via these interfaces. Open standards provide end users with greatly improved competitive choice and functional capability on these systems.
The session will explore examples of IP interfaces for voice dispatch systems and the functions supported, plus give a background on how these apply to many different technologies and can even be adapted for conventional radio applications:
The workshop will cover on the following issues:
- The difference between RoIP and VoIP - how radio systems differ from phone systems
- Implementing one-to-one connections
- Implementing many-to-many connection
- Risk management: Identifying network issues affecting RoIP/VoIP quality; maintenance; and redundancy
- Design elements :- building blocks; calculating network bandwidth requirements
The implications of RoIP for dispatch consoles will be also be discussed: how dispatch console to radio connections can be implemented with RoIP and how RoIP can be used to provide fault tolerant dispatch architectures.
Finally the workshop will look at the impact of new technologies such as IPv6, Wireless Broadband and the switch to Digital Radio on the RoIP landscape.
Les Scott, Manager, System Sales, Zetron
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.
Voice over Internet Protocol (VoIP), is a technology that allows to make voice calls using a broadband Internet connection instead of a regular (or analog) phone line (PSTN).
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
This document provides an overview of VoIP security. It discusses the basics of VoIP security including authentication, authorization, availability, and encryption. It outlines some common attack vectors such as accessing an unsecured local network connection, wireless network, or public network. It also mentions threats from compromising a phone's configuration file or uploading a malicious file. The document summarizes some unconventional VoIP security threats like phishing, caller ID spoofing, eavesdropping, call redirection, and spam over internet telephony.
Voice over IP (VoIP) 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. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service.
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 document provides an overview of Internet Protocol Telephony (VoIP). It discusses how VoIP works by digitizing and compressing voice into packets transmitted over the Internet. It also covers some of the common protocols used, including Session Initiation Protocol (SIP) and H.323, and compares their advantages. Potential applications and challenges of VoIP are also mentioned.
you can be friend with me on orkut
"mangalforyou@gmail.com" : i belive in sharing the knowledge so please send project reports ,seminar and ppt. to me .
This document provides an overview of a project report on Voice over Internet Protocol (VoIP) submitted by two students, Amardeep Singh and Jaswinder Singh, at Chandigarh Engineering College in partial fulfillment of their B-Tech degree in Electronics and Communication Engineering. The report introduces VoIP technology, discusses software and hardware used in the project including Cisco routers and switches, and provides details on configuring an IP phone network with Cisco Call Manager Express including assigning IP addresses via DHCP and configuring phone directory numbers. Future enhancements discussed include integrating VoIP with wireless networks.
VoIP stands for Voice over Internet Protocol. It allows users to make voice calls via a broadband internet connection instead of a regular phone line. There are several VoIP protocols that convert voice into digital signals to transmit over the internet, like SIP and H.323. Users can make VoIP calls from their computers using softphones, or through an Analog Telephone Adapter connected to a regular phone. Setting up a PBX server allows creating a full-featured VoIP phone system. VoIP provides advantages over traditional PSTN phone networks like lower costs and additional features included free of charge.
VOIP Services- We offer High quality Voice over IP services at aggressive pricing.C2C facilitates companies to quickly and economically benefit from the VoIP revolution.
VoIP security involves threats like denial of service attacks, eavesdropping, and quality of service issues. Best practices include using firewalls with application layer gateways or session border controllers, encrypting media and signaling, prioritizing bandwidth for VoIP, and restricting access to call managers through physical and logical security measures. NIST recommends logically separate networks, endpoint encryption, and avoiding vulnerabilities in softphones and wireless networks without encryption.
IP telephony has received interest from many users and organizations as it provides cost savings over traditional phone lines. VoIP saves money by using existing computer networks and IP infrastructure rather than separate phone lines, reducing line charges, feature charges, taxes, and fees. Many organizations currently maintain separate networks for data and voice, but integrating the two using VoIP provides a more cost effective and flexible unified solution.
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.
VoIP uses packet networks to carry voice calls in addition to data. It works by converting analog voice signals to digital data packets which are transmitted over IP networks and reconverted to analog at the receiving end. Key components include IP phones, signaling servers, and protocols like SIP and H.323 which handle call setup and signaling. Quality of service for VoIP depends on factors like packet loss, delay, and jitter which can be managed through queuing and reserving bandwidth for voice traffic.
This document provides an overview and introduction to VoIP and SIP signaling. It discusses key topics such as VoIP architecture and components, the process of a VoIP telephone call including conversion between analog and digital signals and quality of service, SIP architecture including what SIP is, its capabilities and message format, and SIP call flow. The document is intended as a training presentation that includes definitions of terms, descriptions of concepts, diagrams, and quizzes related to VoIP and SIP.
This document discusses VoIP (Voice over Internet Protocol) technology, including its challenges and applications. It covers topics like reliability issues, quality of service, fax transmission, emergency call handling, security concerns, case studies, and the VoIP market. Solutions proposed include improving network infrastructure, prioritizing emergency calls, encrypting VoIP traffic, and segmenting voice and data networks. The VoIP market is projected to grow significantly due to lower costs and the emergence of new communication services.
A NEW SYSTEM ON CHIP RECONFIGURABLE GATEWAY ARCHITECTURE FOR VOICE OVER INTER...csandit
油
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.
Pbx presentation ingate_itexpoeast2014kwader Saudi
油
Enhance employee productivity and reduce communication costs with feature-rich IP telephony solutions from Kwader. With our solutions, your staff can count on effective, unified communications no matter where they are.
KTC scalable IP telephony solutions offer the same high-quality communications whether your enterprise has a few or 100,000 users. Our flexible architecture design offers an unparalleled range of deployment options. Our wide range of resiliency tools minimizes costs and maximizes reliability.
VoIP (Voice over Internet Protocol).pdfOkan YILDIZ
油
VoIP (Voice over Internet Protocol) transmits voice and multimedia content over an internet connection. VoIP allows users to make voice calls from a computer, smartphone, other mobile devices, special VoIP phones and WebRTC-enabled browsers. VoIP is a valuable technology for consumers and businesses, as it typically includes additional features that can't be found on standard phone services. These features include call recording, custom caller ID, and voicemail to e-mail. It is also helpful to organizations as a way to unify communications.
The process works similarly to a regular phone, but VoIP uses an internet connection instead of a telephone company's wiring. VoIP is enabled by a group of technologies and methodologies to deliver voice communications over the internet, including enterprise local area networks or wide area networks.
A VoIP service will convert a user's voice from audio signals to digital data and then send that data through the internet. If another user calls from a regular phone number, the signal is converted back to a telephone signal before reaching that user.
VoIP can also route incoming and outgoing calls through existing telephone networks. However, some VoIP services may only work over a computer or VoIP phone.
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.
7. Softswitch: A programmable network switch that can process the signaling for all
types of packet protocols.
Also called SIP Server, SIP Proxy, Call Manager, VoIP Switch, Call Agent, IP Centrex,
Hosted PBX, User Agent, Gatekeeper (H.323).
According to International Softswitch Consortium, a softswitch should be able to:
1. Control connection services for a media gateway and / or native IP endpoints
2. Select processes that can be applied to a call.
3. Providing routing for a call within the network based on signaling and customer
database information.
4. Transfer the control of the call to another element .
5. Interface to and support management functions such as provisioning, fault, billing,
etc.
The switching technology in a softswitch is in software (hence the name) rather than
in the hardware as with traditional switching center technology.
This software programmability allows it to support IP telephony protocols
(H.323, SIP, MEGACO, etc.).
Examples of Softswitch: Asterisk, 3CX, Volcatel, etc.
http://www.ideapoolonline.com 7
8. In Summary, the two main functions of a Softswitch are:
1. Terminal Control
2. Call Control
Registration: Authentication, directory
entry
Terminal Admission: Permission to make / receive
Control calls
Status: Call disposition
Call Signaling: Address resolution, setup,
tear down
Call Control Call routing
Call accounting
Call detail record
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9. Media Server or Content Server
A computer appliance, ranging from an enterprise class machine with a dedicated
software application providing contents such as voice, video, music, text, graphics.
Two interesting types of media/content:
1. Integrated Messaging
2. Video Server
Integrated Messaging Other Media
Voicemail, Email, Fax Web pages
Text-to-speech, speech-to-text Digital right management
Web-style user interface Music
Google earth
Video Server Countless applications
Basic Cable
Video on Demand: Movie rentals
Pay-per-view, IPTV
PVR functionality
Game Server
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10. Gateways Media Gateway
Gateways are Protocol Converters
Media Gateway converts traditional TDM
to RTP stream and vise versa.
TDM belongs to the PSTN world.
RTP belongs to the IP world.
Media and Signaling gateway could be a
single device or a separate device.
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11. Gateways Signaling Gateway
Gateways are Protocol Converters
Signaling Gateway converts SIP to SS7 and
vice versa.
SIP belongs to the IP world.
SS7 belongs to the traditional PSTN world.
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12. LANs and WANs
LAN: In-building
Ethernet: 10 1000Mbps
Cabling: Cat5, Cat6e twisted pair
MAC Address
Power Over Ethernet
WAN: Between buildings
IP Addresses
Access: DSL, Cable, T1, Fiber, Microwave
Network Protocol: IP, Frame relay, ATM
Carrier Ethernet
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13. Key VoIP Standards
IETF
RFC 3261 SIP
RFC 2327 SDP
RFC 1889 RTP
RFC 0768 UDP
RFC 0791 IP
ITU-T
H.323: Historical
G.700: Voice coding
- G.711: 64kbps PCM
- Many others at lower rates
IEEE802.3
TIA Cable Categories
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14. Where Are We Going?
Broadband IP Dial Tone.
IP-PSTN
-Internet Protocol Packet
Switch Telephone Network.
- IP Packet to any other point
> 1Mbps
Carrier Ethernet
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15. 2. VoIP Architectures and Implementation Choices
Computer-Computer VoIP Over The Internet
Requirements:
- Computer/Mobile Phone/PDA,
- Internet Access.
- Software: Skype, Nimbuzz, IM, .
- Directory, call setup/control.
- Sound card and headset or USB phone.
- Operating system: IP/UDP protocol
stack.
Considerations:
- Free (with internet connection)
- Interoperability: Standard?
- Less user-friendly than PSTN.
- Variable quality of voice
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16. Skype and IM
Proprietary technology
- Skype directory is based on Kazaa
- It uses P2P file sharing
- Good quality voice: Uses 15kbps codec
Works through NATs
- Communications encrypted
- Your computer could relay traffic
Good HMI Human Machine Interface
Yahoo, Google, MSN, AOL, Nimbuzz.
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17. Computer to Phone
Broadband internet connection at one
end.
POTs at other end.
ITSP/VoIP Service provider in the
middle.
Variable voice quality.
E1 connection to circuit-switch PSTN.
Possibly ISP is different from the ITSP.
ITSP is located offshore.
Softphone required to connect to the
ITSP using SIP protocol.
Example of softphone: X-Lite, 3CX, etc.
Example of ITSP: Skype, Belgacom,
etc.
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18. Phone to Phone over the Internet
Broadband internet connection at one
end.
POTs at other end.
ITSP/VoIP Service provider in the
middle.
Variable voice quality.
E1 connection to circuit-switch PSTN.
Possibly, ISP is different from the ITSP.
ITSP is located offshore.
IP phone is required to connect to the
ITSP using SIP protocol.
Example of IP phone: Grandstream,
Cisco, Dlink, etc.
Example of ITSP: Skype, Belgacom,
etc.
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19. Phone to Phone over the Internet
ITSP either supplies the adapter or user
verifies compatible off-the-shelf adapter.
Broadband internet connection at one
end.
POTs at other end.
ITSP/VoIP Service provider in the
middle.
Variable voice quality.
E1 connection to circuit-switch PSTN.
Possibly, ISP is different from the ITSP.
ITSP is located offshore.
POTs connects to the adapter using the
FXS/FXO port.
The adapter acts as FX to VoIP gateway.
Example of ITSP: Skype, Belgacom,
etc.
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20. Managed-IP Telephone Service Provider
All IP end-to-end network.
Radio Access Network could be wireless
- (WiMAX, Wi-Fi, LTE), cable or fibre.
End-to-end Carrier Ethernet Network.
IP network QoS methods.
Voice quality is guarantee.
E1 connection to circuit switched PSTN.
ISP and ITSP same company.
Provider connects with upstream ISP for
long distance call termination and www.
Provider supplies the adapter and
modem.
Example: ipNX
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21. 3. VoIP for Businesses and Organizations
VoIP-Enabled PBX
PBX provides all VoIP functions.
-Call management
- Integrated gateway
- Integrated voicemail
Supports VoIP and traditional phones
-Phased implementation of VoIP
in-building.
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23. Hosted PBX
Softswitch at Service Provider
-SIP messages over network
- Customer configuration via web portal
- IVR, ACD hosting too
Strategic Advantages
-Professional implementation
- Scalability, flexibility
- Security, reliability
Financial advantages
-Lease vs. buy: capital, maintenance
- Reduce staffing
Disadvantages
-Lack of control
- Feature set? Capacity?
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24. IP Centrex
Hosted softswitch service from the Phone
Company
-Huge Centrex customer base
Higher level of trust
-Professionalism, longevity
Multilocation benefits
-Uniform services, support
Bundled services
-Access circuit plus telephony
Seamless services
-Home-to-office VPN, telephony
Features and pricing
-Per user, per month
- Local number, DID
- Desktop, web collaboration
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25. Open-sourced IP-PBX Software
Free code updated by users
Asterisk: One of the Oldest
-IP PBX plus signaling server,
media gateway, analog/TDM PBX, IVR,
messaging server
Supported commercially
-30 50% cost of proprietary
- Asterisk (Digium)
- sipX, SIPfoundry (Pingtel)
- Others: vovida, Freeswitch
Disadvantages
-May be difficult to implement
- Requires skilled technicians
- Lack of management/support tools
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26. SO/HO IP Phone Features and Uses
Small / home office
All-in-one device
-Fast processor, jitter buffer
- Linux OS, browser, software
- SIP, MGCP
- Ethernet switch, PoE, PPPoE
- Bluetooth
- Router, DHCP, NAT, firewall
- VPN, QoS support
Dialing features dial by:
- Phone number, SIP address,
IP address, directory, voice recognition,
softkeys, presence/location notice,.
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