VoIP provides voice calls over the internet using packet switching rather than traditional circuit switching. The document discusses the technology behind VoIP, including standards like SIP and H.323. While VoIP is still developing, services are appearing that offer VoIP using these standards. The document also outlines issues that have arisen with VoIP implementation and research being done to address these issues, focusing on a case study from the University of Southampton.
This document provides a summary of integrating IP telephony into the public switched telephone network (PSTN) environment. It discusses the evolution of PSTN from analog to digital networks and the emergence of time division switching. It then outlines the architecture of a soft switch solution for integration, including edge, core, control, and application layers. Finally, it discusses existing practices for integration, such as replacing tandem exchanges, and considers the specific context of integrating IP networks in Bangladesh.
The document discusses the evolution and integration of IP telephony with traditional PSTN networks over six stages. It describes alternatives to legacy PBX systems such as using IP telephony to replace inter-building connections or fully replacing the PBX. The document also discusses concepts like least-cost routing, IP telephony gateways, packet-based switches, and integrating VoIP with existing PBX systems.
The document discusses how IP telephony can provide voice communication services to rural areas in Bangladesh. It describes how IP-based networks are well-suited for quickly implementing telephone infrastructure in rural regions. Specifically, the document outlines various IP network architectures that could be used to deliver rural telephony services and discusses technical aspects of implementing VoIP systems, including considerations around reliability, quality of service, emergency calls, and security.
This document discusses VoIP (Voice over Internet Protocol) techniques and challenges. It begins by introducing VoIP as an alternative to traditional PSTN telephone networks that transmits voice over internet and packet-switched networks more cheaply. However, ensuring high quality of service (QoS) with factors like jitter, packet delay/loss, and bandwidth allocation presents major challenges for VoIP systems. The document goes on to describe how VoIP works by carrying voice in RTP packets within IP packets and discusses challenges to VoIP performance and QoS from system capacity, packet loss, delay, jitter, echo, and security.
Voice over IP (VoIP) is a technology that allows users to make voice calls using a broadband Internet connection instead of a regular phone line. It converts the voice signal from an analog signal to digital data that can be transmitted over the Internet or IP networks. Popular protocols used for VoIP include SIP, H.323, and Skype. VoIP saw widespread adoption among consumers as broadband access became more available and VoIP services offered unlimited calling for a flat monthly fee. Businesses also migrated phone systems to VoIP to reduce costs. Challenges of VoIP include quality of service, compatibility with analog phones, emergency call support, and security.
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.
Customized IVR Implementation Using Voicexml on SIP (Voip) Communication Plat...IJMER
This document discusses implementing customized interactive voice response (IVR) applications using VoiceXML on a VoIP communication platform. It provides an overview of key concepts like VoIP, SIP, IPPBX, and VoiceXML. VoIP allows voice calls over IP networks using protocols like SIP. An IPPBX functions similar to a traditional PBX but routes calls over an IP network. VoiceXML is used to build voice dialog applications with features like speech recognition, prompts, and grammars. The document proposes using a VoiceXML browser integrated with an Asterisk gateway and IPPBX to enable voice applications with speech input/output that can be customized for different organizations.
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 provides a summary of integrating IP telephony into the public switched telephone network (PSTN) environment. It discusses the evolution of PSTN from analog to digital networks and the emergence of time division switching. It then outlines the architecture of a soft switch solution for integration, including edge, core, control, and application layers. Finally, it discusses existing practices for integration, such as replacing tandem exchanges, and considers the specific context of integrating IP networks in Bangladesh.
The document discusses the evolution and integration of IP telephony with traditional PSTN networks over six stages. It describes alternatives to legacy PBX systems such as using IP telephony to replace inter-building connections or fully replacing the PBX. The document also discusses concepts like least-cost routing, IP telephony gateways, packet-based switches, and integrating VoIP with existing PBX systems.
The document discusses how IP telephony can provide voice communication services to rural areas in Bangladesh. It describes how IP-based networks are well-suited for quickly implementing telephone infrastructure in rural regions. Specifically, the document outlines various IP network architectures that could be used to deliver rural telephony services and discusses technical aspects of implementing VoIP systems, including considerations around reliability, quality of service, emergency calls, and security.
This document discusses VoIP (Voice over Internet Protocol) techniques and challenges. It begins by introducing VoIP as an alternative to traditional PSTN telephone networks that transmits voice over internet and packet-switched networks more cheaply. However, ensuring high quality of service (QoS) with factors like jitter, packet delay/loss, and bandwidth allocation presents major challenges for VoIP systems. The document goes on to describe how VoIP works by carrying voice in RTP packets within IP packets and discusses challenges to VoIP performance and QoS from system capacity, packet loss, delay, jitter, echo, and security.
Voice over IP (VoIP) is a technology that allows users to make voice calls using a broadband Internet connection instead of a regular phone line. It converts the voice signal from an analog signal to digital data that can be transmitted over the Internet or IP networks. Popular protocols used for VoIP include SIP, H.323, and Skype. VoIP saw widespread adoption among consumers as broadband access became more available and VoIP services offered unlimited calling for a flat monthly fee. Businesses also migrated phone systems to VoIP to reduce costs. Challenges of VoIP include quality of service, compatibility with analog phones, emergency call support, and security.
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.
Customized IVR Implementation Using Voicexml on SIP (Voip) Communication Plat...IJMER
This document discusses implementing customized interactive voice response (IVR) applications using VoiceXML on a VoIP communication platform. It provides an overview of key concepts like VoIP, SIP, IPPBX, and VoiceXML. VoIP allows voice calls over IP networks using protocols like SIP. An IPPBX functions similar to a traditional PBX but routes calls over an IP network. VoiceXML is used to build voice dialog applications with features like speech recognition, prompts, and grammars. The document proposes using a VoiceXML browser integrated with an Asterisk gateway and IPPBX to enable voice applications with speech input/output that can be customized for different organizations.
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 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.
VoIP is an emerging technology that uses the internet to transmit phone calls rather than traditional telephone networks. It has the potential to significantly lower phone costs for consumers while improving quality. VoIP services are already commercially available and most are low cost or free. However, the existing telecommunications regulations only apply to traditional phone networks, so a new regulatory framework is needed to address VoIP. Both incumbent phone companies and new internet-based companies see opportunities in VoIP, but there are debates around how it should be regulated.
IRJET-Identifying Disaster Area using Wireless TechnologyIRJET Journal
1. The document discusses using existing WiFi networks and technology like Asterisk to enable communication in disaster areas by connecting mobile phones and intercom systems over IP networks at no cost.
2. It aims to build a common IP-PBX system using Asterisk server to connect traditional analog phones, softphones, and VoIP phones. Wireless access points would connect smartphones to the VoIP network.
3. Key technologies discussed include VoIP, Asterisk software which supports protocols like SIP, and using WiFi networks in either peer-to-peer or infrastructure modes to enable emergency communication when conventional networks are disrupted.
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.
LTE provides significantly higher data throughput and lower latency than previous mobile network technologies. This allows for improved quality of experience for users accessing real-time services like voice calls, gaming, and video. LTE will also enable new high bandwidth services like HD video and multi-user interactive gaming. Alcatel-Lucent's LTE solution aims to meet users' quality of experience expectations and reduce costs for network operators by introducing a flat IP architecture.
The document discusses Internet telephony and Voice over Internet Protocol (VoIP). It defines IP telephony as using Internet protocol to exchange voice, fax, and other information traditionally carried over telephone networks. VoIP aims to standardize IP telephony. The document then describes how a VoIP call is completed by digitizing, compressing, and transmitting voice data over the Internet in packets before reassembling it at the destination. It notes benefits of IP telephony include potential cost savings compared to traditional telephone networks.
The document discusses Voice over IP (VoIP) and IP telephony. It explains that VoIP allows phone calls to be made over an IP network like the internet instead of the traditional public switched telephone network (PSTN). VoIP offers cost savings compared to PSTN and enables additional features like video calls and mobility. The document also discusses when companies should consider replacing their private branch exchange (PBX) phone system with an IP telephony system using VoIP.
The document discusses considerations for service providers offering Voice over IP (VoIP) services. Key points include:
- VoIP allows providers to offer voice, video, and other multimedia services over a single IP-based network.
- Quality of service is important to deliver reliable voice services. Technologies like DiffServ and MPLS traffic engineering help prioritize voice traffic.
- Signaling protocols like SIP and H.323 set up calls between endpoints. Client-server protocols interact with call controllers and gateways.
- Foundry Networks' routers and switches enable scalable VoIP solutions through features like traffic prioritization, high availability, and security.
This document summarizes a student project on Voice over IP (VoIP) quality of service. It discusses how VoIP works by converting analog speech to digital packets sent over the Internet. It then covers current Internet limitations for real-time applications like VoIP. It evaluates scheduling algorithms like FIFO, priority queueing, and weighted fair queueing. The document outlines simulating these algorithms in OPNET and analyzing results. Based on this, it proposes a new algorithm using priority queueing for real-time traffic and weighted fair queueing with dynamic weights for other traffic. Simulation results show the proposed algorithm meeting quality of service requirements for different traffic classes.
Atento needed to connect call centers in Spain with customer service representatives in South America to reduce costs. RAD's Vmux voice compression gateways were deployed to compress voice traffic over data links at a rate of 5.3 kbps, about half the rate of VoIP. This allowed Atento to eliminate expensive international E1 lines and reduce operating expenses. The modular RAD solution was simple to install and deploy new services and desktops quickly.
This document discusses implementing Voice over IP (VoIP) and IP Multimedia Subsystem services over WiMAX wireless networks. It addresses introducing VoIP and multimedia transmission over wireless, using soft switching for compatibility with WiMAX. It also discusses challenges like ensuring voice quality, security, and E911 support. Finally, it explores services like video on demand that WiMAX networks can provide using IP Media Subsystem technologies.
The document provides an overview of the IP Multimedia Subsystem (IMS) architecture, protocols, and services. Key points include:
- IMS provides an integrated architecture for multimedia services over different access networks through the use of IP.
- It allows for person-to-person and person-to-content communications using voice, text, pictures and video on both wireless and fixed networks.
- The IMS architecture includes the Call Session Control Function (CSCF), Home Subscriber Services (HSS), Application Servers, and other network elements that provide services like authentication, authorization, charging and quality of service.
- IMS supports multimedia applications and services like presence, instant messaging, push-
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.
The document discusses fax over IP (FoIP) and SIP trunking. It provides an overview of FoIP technologies like T.38 and G.711, factors that influence FoIP success like network impairments, and considerations for choosing a SIP trunk provider such as whether they support T.38 and have proven interoperability with key equipment.
Nowadays VoIP technologies have taken the upper hand offering many advantages compared to the traditional telephone network, but what are the security risks involved when voice and data networks come together. In this presentation, we will identify and evaluate these different security risks and their countermeasures both from a defensive as offensive position.
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 discusses VoIP in mobile communication. It provides an overview of how VoIP works using packet switching instead of circuit switching. It then discusses mobile communication standards like GSM and 3G. It explores how VoIP can be used with wireless phones and whether VoIP is likely to be adopted by mobile carriers. While mobile VoIP is growing, the document argues that mobile carriers will not adopt VoIP themselves due to bandwidth constraints and lack of technological advantages over existing standards like GSM.
The document concludes that VoIP subscriber growth is entering the mainstream in the US, especially for residential and business use over the next few years, though full migration will take much longer as traditional phone networks still dominate mobile communication globally.
Practical Fundamentals of Voice over IP (VoIP) for Engineers and TechniciansLiving Online
This manual provides solid practical advice on application, implementation and, most importantly, troubleshooting Voice Over IP (VOIP) systems.
MORE INFORMATION: http://www.idc-online.com/content/practical-fundamentals-voice-over-ip-voip-21?id=151
This document provides an overview of the Internet and its applications. It discusses various topics including Internet architectures (Internet, intranet, extranet), communication applications (email, messaging, e-commerce, voice over IP), virtual private networks, WAN technologies, and audio/video streaming and conferencing.
The document discusses the evolution of mobile devices and wireless communications. It provides an overview of how mobile devices have advanced from basic voice-only capabilities to integrated voice and data. It also outlines the evolution of wireless technologies and standards such as GSM, UMTS, HSPA, LTE, and 5G. Finally, it provides background on communications and telecommunications definitions and the radio spectrum used for wireless transmissions.
Burbank Middle School in Houston, Texas has been a recognized school for using technology in teaching and learning for the past 3 years based on assessments using the Texas STaR Chart. The STaR Chart evaluates 4 domains: Teaching & Learning, Educator Preparation & Development, Leadership/Administration/Support, and Infrastructure. Over 3 years of assessments, Burbank showed strengths in Teaching & Learning but weaknesses in Infrastructure and Leadership. To continue advancing technology use, future recommendations include ongoing technology planning, professional development, access to technology for all students, and use of digital tools meeting standards.
Customer reference programs should have strategic goals that benefit both parties, not just one. To be effective, the program goals should align with the business objectives and create a structured framework for curating and sharing content with customers. The success of the program can be measured by tracking customer progression through the framework, linking to sales data, and ensuring all participants are happy.
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.
VoIP is an emerging technology that uses the internet to transmit phone calls rather than traditional telephone networks. It has the potential to significantly lower phone costs for consumers while improving quality. VoIP services are already commercially available and most are low cost or free. However, the existing telecommunications regulations only apply to traditional phone networks, so a new regulatory framework is needed to address VoIP. Both incumbent phone companies and new internet-based companies see opportunities in VoIP, but there are debates around how it should be regulated.
IRJET-Identifying Disaster Area using Wireless TechnologyIRJET Journal
1. The document discusses using existing WiFi networks and technology like Asterisk to enable communication in disaster areas by connecting mobile phones and intercom systems over IP networks at no cost.
2. It aims to build a common IP-PBX system using Asterisk server to connect traditional analog phones, softphones, and VoIP phones. Wireless access points would connect smartphones to the VoIP network.
3. Key technologies discussed include VoIP, Asterisk software which supports protocols like SIP, and using WiFi networks in either peer-to-peer or infrastructure modes to enable emergency communication when conventional networks are disrupted.
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.
LTE provides significantly higher data throughput and lower latency than previous mobile network technologies. This allows for improved quality of experience for users accessing real-time services like voice calls, gaming, and video. LTE will also enable new high bandwidth services like HD video and multi-user interactive gaming. Alcatel-Lucent's LTE solution aims to meet users' quality of experience expectations and reduce costs for network operators by introducing a flat IP architecture.
The document discusses Internet telephony and Voice over Internet Protocol (VoIP). It defines IP telephony as using Internet protocol to exchange voice, fax, and other information traditionally carried over telephone networks. VoIP aims to standardize IP telephony. The document then describes how a VoIP call is completed by digitizing, compressing, and transmitting voice data over the Internet in packets before reassembling it at the destination. It notes benefits of IP telephony include potential cost savings compared to traditional telephone networks.
The document discusses Voice over IP (VoIP) and IP telephony. It explains that VoIP allows phone calls to be made over an IP network like the internet instead of the traditional public switched telephone network (PSTN). VoIP offers cost savings compared to PSTN and enables additional features like video calls and mobility. The document also discusses when companies should consider replacing their private branch exchange (PBX) phone system with an IP telephony system using VoIP.
The document discusses considerations for service providers offering Voice over IP (VoIP) services. Key points include:
- VoIP allows providers to offer voice, video, and other multimedia services over a single IP-based network.
- Quality of service is important to deliver reliable voice services. Technologies like DiffServ and MPLS traffic engineering help prioritize voice traffic.
- Signaling protocols like SIP and H.323 set up calls between endpoints. Client-server protocols interact with call controllers and gateways.
- Foundry Networks' routers and switches enable scalable VoIP solutions through features like traffic prioritization, high availability, and security.
This document summarizes a student project on Voice over IP (VoIP) quality of service. It discusses how VoIP works by converting analog speech to digital packets sent over the Internet. It then covers current Internet limitations for real-time applications like VoIP. It evaluates scheduling algorithms like FIFO, priority queueing, and weighted fair queueing. The document outlines simulating these algorithms in OPNET and analyzing results. Based on this, it proposes a new algorithm using priority queueing for real-time traffic and weighted fair queueing with dynamic weights for other traffic. Simulation results show the proposed algorithm meeting quality of service requirements for different traffic classes.
Atento needed to connect call centers in Spain with customer service representatives in South America to reduce costs. RAD's Vmux voice compression gateways were deployed to compress voice traffic over data links at a rate of 5.3 kbps, about half the rate of VoIP. This allowed Atento to eliminate expensive international E1 lines and reduce operating expenses. The modular RAD solution was simple to install and deploy new services and desktops quickly.
This document discusses implementing Voice over IP (VoIP) and IP Multimedia Subsystem services over WiMAX wireless networks. It addresses introducing VoIP and multimedia transmission over wireless, using soft switching for compatibility with WiMAX. It also discusses challenges like ensuring voice quality, security, and E911 support. Finally, it explores services like video on demand that WiMAX networks can provide using IP Media Subsystem technologies.
The document provides an overview of the IP Multimedia Subsystem (IMS) architecture, protocols, and services. Key points include:
- IMS provides an integrated architecture for multimedia services over different access networks through the use of IP.
- It allows for person-to-person and person-to-content communications using voice, text, pictures and video on both wireless and fixed networks.
- The IMS architecture includes the Call Session Control Function (CSCF), Home Subscriber Services (HSS), Application Servers, and other network elements that provide services like authentication, authorization, charging and quality of service.
- IMS supports multimedia applications and services like presence, instant messaging, push-
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.
The document discusses fax over IP (FoIP) and SIP trunking. It provides an overview of FoIP technologies like T.38 and G.711, factors that influence FoIP success like network impairments, and considerations for choosing a SIP trunk provider such as whether they support T.38 and have proven interoperability with key equipment.
Nowadays VoIP technologies have taken the upper hand offering many advantages compared to the traditional telephone network, but what are the security risks involved when voice and data networks come together. In this presentation, we will identify and evaluate these different security risks and their countermeasures both from a defensive as offensive position.
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 discusses VoIP in mobile communication. It provides an overview of how VoIP works using packet switching instead of circuit switching. It then discusses mobile communication standards like GSM and 3G. It explores how VoIP can be used with wireless phones and whether VoIP is likely to be adopted by mobile carriers. While mobile VoIP is growing, the document argues that mobile carriers will not adopt VoIP themselves due to bandwidth constraints and lack of technological advantages over existing standards like GSM.
The document concludes that VoIP subscriber growth is entering the mainstream in the US, especially for residential and business use over the next few years, though full migration will take much longer as traditional phone networks still dominate mobile communication globally.
Practical Fundamentals of Voice over IP (VoIP) for Engineers and TechniciansLiving Online
This manual provides solid practical advice on application, implementation and, most importantly, troubleshooting Voice Over IP (VOIP) systems.
MORE INFORMATION: http://www.idc-online.com/content/practical-fundamentals-voice-over-ip-voip-21?id=151
This document provides an overview of the Internet and its applications. It discusses various topics including Internet architectures (Internet, intranet, extranet), communication applications (email, messaging, e-commerce, voice over IP), virtual private networks, WAN technologies, and audio/video streaming and conferencing.
The document discusses the evolution of mobile devices and wireless communications. It provides an overview of how mobile devices have advanced from basic voice-only capabilities to integrated voice and data. It also outlines the evolution of wireless technologies and standards such as GSM, UMTS, HSPA, LTE, and 5G. Finally, it provides background on communications and telecommunications definitions and the radio spectrum used for wireless transmissions.
Burbank Middle School in Houston, Texas has been a recognized school for using technology in teaching and learning for the past 3 years based on assessments using the Texas STaR Chart. The STaR Chart evaluates 4 domains: Teaching & Learning, Educator Preparation & Development, Leadership/Administration/Support, and Infrastructure. Over 3 years of assessments, Burbank showed strengths in Teaching & Learning but weaknesses in Infrastructure and Leadership. To continue advancing technology use, future recommendations include ongoing technology planning, professional development, access to technology for all students, and use of digital tools meeting standards.
Customer reference programs should have strategic goals that benefit both parties, not just one. To be effective, the program goals should align with the business objectives and create a structured framework for curating and sharing content with customers. The success of the program can be measured by tracking customer progression through the framework, linking to sales data, and ensuring all participants are happy.
Ice Blue Sky is a marketing and communications firm that helps technology companies align their marketing messaging with how customers purchase technologies. The firm translates technical features into benefits for customers by showing how products can solve problems. Ice Blue Sky works with companies to understand customer needs and demonstrate what's in it for them beyond just the product's functions. They offer more than just communications services and encourage interested companies to contact them.
The document provides guidance for evangelists on spreading the message of truth effectively. It advises that time is running out to share the message before obstacles are put in place. Evangelists must work aggressively but humbly, learning from Jesus, and watching out for Satan's plans to distract and divide them. It suggests focusing first on less controversial truths to gain people's trust and confidence before introducing more difficult topics like the Sabbath. Evangelists are warned not to publicly advertise their plans to avoid opposition from enemies, but to work strategically in secret like wise generals. Tact and careful planning of what to say is advised over always sharing all truths or practicing deception.
The document discusses the benefits of meditation for reducing stress and anxiety. Regular meditation practice can help calm the mind and body by lowering heart rate and blood pressure. Making meditation a part of a daily routine, even if just 10-15 minutes per day, can offer improvements to mood, focus, and overall well-being over time.
This 3 page document is a book written and illustrated by a kindergarten class about mothers. It describes different activities that mothers do with their children like cooking, reading stories before bed, and helping with homework. The book aims to celebrate mothers and the important role they play in their children's lives.
Taking Impressions with Mandibular Tori Interference using Triotray by Dr Gra...Triodent
Triotray, with its unique side tabs plastic tipped for patient comfort, is a rigid and accurate dual arch tray that will consistently take successful posterior impressions, saving time and eliminating frustration.
Do cavity on upper first molar by dr simon mc donaldTriodent
The document describes the clinical case of restoring a cavity on an upper first molar tooth using a V3 Tab-Matrix system. The dentist prepares the cavity, places a glass ionomer base, positions the V3 Tab-Matrix with a contra-angle tab and Wave-Wedge, and then places composite filling material. After curing the composite, the dentist removes the matrix and evaluates the proximal contour and marginal ridge form before minimal finishing and achieving the final restored tooth.
MYOB ClientConnect is a customer relationship management (CRM) system that integrates with Microsoft Outlook. It allows users to manage contacts and sales opportunities, provide customized reports, and automate workflows. The system benefits businesses by helping generate more sales, improving customer relations, and centralizing data management. There are three subscription-based versions available ranging from $119 to $799 per user annually. It requires installation on the server and clients and must be activated within 30 days.
The document is about ISM Sports Day held in February 2010. It introduces a sports team called the Flashing Fireflies that will be participating in the sports day events.
Este álbum de fotografias contém imagens de momentos especiais e pessoas importantes para St. As fotos mostram lembranças felizes do passado que St quer guardar e se lembrar.
IRJET- Campus-Wide Internet Telephony Design and Simulation using Voice over ...IRJET Journal
This document discusses the design and simulation of a Voice over Internet Protocol (VoIP) system for Adamawa State University in Nigeria using Cisco Packet Tracer. VoIP allows voice calls to be placed over an IP network like the internet rather than a traditional phone network. The proposed VoIP system would allow users across the university's campus to communicate freely using IP phones. The author conducted several simulations of the network architecture in Cisco Packet Tracer to develop a prototype VoIP system for the university. This would provide more flexible communication and help increase information sharing across the university's departments and offices by integrating them into a single network.
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.
The document provides an overview of Voice over IP (VoIP) including its benefits and requirements. VoIP allows phone calls to be made over an IP network like the internet rather than the traditional public switched telephone network. It provides benefits like reduced costs and integrated services. However, deploying VoIP requires addressing requirements around services, quality of service, security, billing and network interconnection to provide equivalence to the PSTN. The document also discusses protocols used for VoIP like OSI and layers of the OSI model.
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.
This document provides an overview of Voice over Internet Protocol (VoIP) technology. It defines VoIP as using broadband internet connections to make phone calls over the internet. The document outlines the agenda which includes what VoIP is, reasons to migrate to VoIP, general VoIP configuration, advantages, and disadvantages. It also discusses how VoIP is an emerging technology that is important for network-centric warfare.
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.
This document discusses softswitches, which are central devices that connect phone calls entirely through software running on computer systems, replacing physical switchboards. It provides background on softswitches and their advantages over hardware switches, noting they can reduce costs, improve services, and facilitate migration to IP networks. Key elements of a softswitch architecture are described, including media gateways and call agents. Benefits of softswitch architecture for wireless networks are outlined, such as enabling distributed switching to reduce costs compared to centralized hardware switches.
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.
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.
A Low-Cost Telephony System For Small Medium Scale Businesses In Nigeria An ...Michele Thomas
This document discusses implementing VoIP (Voice over Internet Protocol) on a local area network (LAN) using a mini SIP server to provide telephony services for small and medium-sized businesses in Nigeria. It describes how traditional telephone networks work and how VoIP converts voice calls to digital signals that can be transmitted over IP networks. The study simulated a VoIP system on a school LAN using a miniSIP server and Zoiper softphone client to allow voice and video calls between connected devices. Testing showed the system successfully established calls within the LAN. Implementing VoIP in this way provides a cheaper alternative to traditional telephone networks for communication within an organization.
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.
VoIP (Voice Over IP) allows users to make phone calls using an Internet connection rather than a traditional phone line. It works by converting voice signals into digital data packets which are transmitted over the Internet or other IP-based networks. Common protocols used for VoIP include UDP, RTP, and SIP. While VoIP provides advantages like lower costs, it also faces challenges of packet loss, latency, jitter, and firewall restrictions that can impact call quality.
Madhumita Routray presented on Internet Protocol Telephony. The presentation covered:
1. IP telephony uses IP networks to transmit voice traffic instead of traditional telephone networks. It has lower costs and provides more features.
2. The architecture includes end devices, gateways, and gatekeepers. Protocols like H.323 and SIP are used to connect the different components.
3. IP telephony works by digitizing voice into packets that are transmitted over IP networks and reassembled at their destination. There are challenges around quality of service and integrating with traditional telephone networks.
SIP Trunking - The cornerstone of unified communicationsJake Weaver
SIP trunks perform some key call and session control and management functions, and serve in place of traditional access lines and trunks.ix Typically provisioned by carriers over T1 trunks, carrier-based SIP trunk services interface between carriers’ nodes and SIP-enabled customer premises equipment, such as a SIP gateway or IP PBX.x SIP Trunks also interconnect customer sites with hosted VOIP/IP centrex services, cloud/software as a service (SaaS) applications, and facilitate customer connectivity with IMS-based applications, like single number service (see sidebar). At the carrier, SIP Trunks interconnect to IP-based WAN services, VOIP/multimedia-enabled network nodes and from these, to the public switched telephone network (PSTN).
AN OVERVIEW OF VOICE OVER INTERNET PROTOCOL (VOIPSean Flores
This document discusses Voice over Internet Protocol (VoIP) including its protocols, security issues, benefits, and challenges. It begins by introducing VoIP and describing its basic operation and advantages like lower costs. It then covers specific VoIP protocols like SIP and H.323. The document analyzes VoIP considerations like delay, jitter, packet loss, and discusses how these issues can affect call quality. It also provides an overview of VoIP technologies and their benefits for businesses. Finally, it presents a case study on assessing network readiness for VoIP deployment.
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.
In this white paper, VoIP for Beginners, you’ll be introduced to how VoIP works.
Discover what occurs when a VoIP call is placed and received
Understand the key technical terms and learn the issues that affect bandwidth and call quality Learn three issues to consider when defining VoIP call quality
VOIP, or Voice over Internet Protocol, allows users to make phone calls using an Internet connection instead of a regular phone line. It works by converting analog audio signals into digital data that can be transmitted over the Internet. VOIP provides the ability to make free phone calls through a standard Internet connection and has emerged as an innovative technology that can transform phone systems globally.
This document discusses a fraud monitoring system for voice over internet protocol (VoIP) telephony. It begins with an introduction to VoIP and defines fraud. It then discusses the history of VoIP and how VoIP connections work. Key points discussed include quality of service requirements, protocols used in VoIP like SIP and H.323, and security challenges like dynamic addressing and firewalls. The document examines how a fraud management system could address these security issues to help secure VoIP networks.
Welcome to International Journal of Engineering Research and Development (IJERD)IJERD Editor
The document summarizes research on using Voice over Internet Protocol (VoIP) for voice communication over wireless networks. VoIP digitizes voice and transmits it over the internet using protocols like SIP and RTP. Factors that affect VoIP quality include delay, jitter, and packet loss. The document discusses how VoIP can provide lower communication costs compared to traditional phone networks and allow optimized functionality like video/audio chats. Routing in VoIP networks aims to ensure stability through well-connected nodes and standardized transmission protocols to minimize packet loss and jitter. Services provided by VoIP include PC-to-PC, PC-to-phone, and phone-to-phone calls over IP networks. Potential drawbacks are reliance on
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2. singular connection is thus achieved between the congested and quickest route. Figure 2 outlines this
two clients via this route and voice packets can now process and shows two clients connecting over the
be transmitted. Figure 1 shows a break down of internet.
phone number into its many parts showing how a
call would be established between 2 clients. Call CLIENT DISCOVERY
costs can be implied by simply finding out which VoIP SERVERS
LATAs have been traversed and applying charges
appropriately [5].
INTERNET
2.1. Telephony into VoIP
VoIP is based heavily on the already existent
structure of the worldwide PSTN, however the
active environment is the internet and thus VoIP has VoIP CALL – DIRECT CONNECTION
been tuned to use existing network protocols where
CLIENT 1 CLIENT 2
available. Like the PSTN network a user will be
connected to a local exchange (server) which in turn Figure 2 – VoIP Call example (discovery and connection)
is connected to other servers around the world.
This decentralized architecture is ideal for end to
These servers are able to communicate freely with
end connections of only two users, however
each other in order to find and connect users [5].
connection and management of conference calls
VoIP has two main deployment methods based
becomes more of a challenge. Making multipoint
upon protocols from different developers. The ITU-T
calls involves using IP multicast to transmit data to
recommendation H.323 [3] follows a client server
many users, which means that users must be able
architecture much like the worldwide PSTN. Clients
to transmit and receive multicast packets at their
interact both for data transport and control with a
location.
small number of servers which coordinate and
control the session. The IETF recommends the
Session Initiation Protocol (SIP) [4] which is a highly
decentralized architecture where servers are only
used to locate users. A peer to peer link over the
internet can then be established between the users
without the need for an expensive powerful server.
3. THE SESSION INITIATION PROTOCOL
(SIP)
SIP (Session initiation protocol) is an Internet
standard specified by the Internet Engineering Task
Force (IETF) in RFC 2543 [4]. SIP is used to initiate,
manage, and terminate interactive sessions
between one or more users on the Internet. SIP
borrows heavily from HTTP and the e-mail protocol Figure 3 – Structure of SIP
SMTP, providing scalability, extensibility, flexibility,
and capabilities for creation of new services. As a While servers are required to carry out some of the
result SIP is increasingly used for Internet telephony more complex SIP features such as transcoding, it
signalling, in gateways, PC phones, softswitches, is possible set up point to point or multicast
and softphones, however is not limited to Internet conference calls without the need for a server. SIP
telephony and can be used to initiate and manage has been designed specifically to allow clients to
any type of session, including video, interactive make use of IP packets for both control and data
games, and text chat. SIP takes advantage of the transport within calls.
underlying technology of the internet, harnessing
A generic SIP call involves a SIP User Agent (UA)
this where possible so as to decentralize any
locating a user on a registrar server (VoIP server)
dependencies on the SIP server. A good example of
and then issuing an invitation to them via a proxy
which is how users are connected over a SIP
server making use of any redirect servers where
network: Unlike PSTN once the two users are
appropriate. A successful SIP invitation consists of
located the call is not connected via the servers or
two messages: INVITE followed by an ACK. The
the route taken in order to find the users. The
INVITE message contains a session description
internet already contains a route optimisation
from the UA containing information on which type of
framework at the packet level and thus users are
media to caller wishes to use and can accept for the
connected direct to each other using a peer to peer
call. Media types, often referred to as codecs
link. By default packets will traverse the least
3. included many such as GSM and the ITU codecs, these being those which establish and negotiate the
some of which are already in use on mobile phone calling properties and media transmission types in
networks and in other commercial voice use. H.450 defines a generic functional protocol on
applications. This capability enables SIP to take full top of H.225 for all supplementary services and
advantage of current technology and be integrated provides the only abstraction layer to H.323 where
where possible. extra services can be harnessed in a call.
Being based heavily on the SMTP and HTTP
protocols, SIP adopts many of the methods and
usability from these. None more so than user
location, as well as each user having a unique
number, users can also have a user@host.domain
address which is simply aliased to their number.
Finally SIP also provides a Session Announcement
Protocol (SAP) and the Session Description
Protocol (SDP) which support the establishment of
multiparty conferencing sessions. SDP defines the
description of multimedia sessions, while SAP
enables periodic multicasting of information about
active sessions. Together these enable third party
users to join an already established session within a
given time frame. Figure 4 – Structure of H.323
Figure 3 depicts the overall IETF SIP protocol suite H.323 can have a number of servers to perform
and the many extensions available, with space left different tasks depending on the scenario. Typically
for many more which are being worked upon by the an H.323 gatekeeper (GK) performs many of tasks
IETF as internet drafts. SIP itself simply provides a equivalent to the SIP proxy server providing address
small number of text based messages to be translation, RAS control, call redirection and
exchanged in separate transactions between the resource management. H.323 can also create
SIP peer entities. The session itself is described at decentralised point to point links to users for use in
two levels. The SIP protocol contains the parties’ calls however the gatekeeper hander the
addresses and protocol processing features (media initialisation and termination of the session rather
types), with the body containing SDP which is a than the individual clients. In decentralised calling
structured, text-based media description format. mode each client must also act as multipoint
Since the message body is transparent to SIP any processor and be able to process media streams,
type of SDP can be transferred, thus not limiting SIP including multicast.
to use in VoIP, but opening it up to use in any
To enable advanced features such as conference
session based application. SIP extensions such as
calling in H.323 further servers must be used to
event notification (RFC 3265), session update (RFC
establish and manage connections between multiple
3311), call transfer and call holding can then be
users. A multipoint controller (MC) establishes an
applied to complete the SIP core framework and
H.245 control connection to each user for
use in VoIP.
negotiation of media communication types. The
4. H.323 multipoint processor (MP) is then able to decode
and retransmit the streams as required. The H.323
H.323 defines system aspect requirements for MC component is responsible for selecting unicast
multimedia communication systems over a packet or multicast media transmission and for choosing
switching network. This includes registration, network/transport addresses.
admission and status (RAS or RTP/RTCP) control,
call setup as defined in H.225.0 and call setup and To establish a call using H.323 the II.323 call
signalling as defined in H.245. H.225.0 defines an signalling procedure has to be carried out to
alias type for carrying any standard Uniform establish valid H.245 connections via the
Resource Locator (URL) [11]. H.323 version 4 [3] gatekeeper. The II.323 call signalling procedure
introduced an H.323-specific URL, which may be begins when an originating H.323 client issues an
used to resolve the address of a network entity to admission request (ARQ) to local gatekeeper in its
which H.323 calls may be directed. Like SIP H.323 domain. When the corresponding confirmation
also supports many audio and video codecs for use message (ACF) is received the call setup procedure
in calls, as well as real time media transport continues with a SETUP and CONNECTION
protocols (RTP and RTCP). message exchange. Upon successful establishment
of a call the clients follow the H.245 capability
Figure 4 depicts the structure of the H.323 protocol exchange procedure to open media channels which
suite showing the compulsory objects in dark tan, both clients are able to support. In later versions (3
4. upwards) of H.323 clients are able to reduce gives a comparison between the many codecs with
signalling overhead by using the Fast Connection bit rate (quality) and bandwidth required for each to
procedure. This “FastStart” procedure is included as be used. With the operation of VoIP on a large scale
an element in the SETUP message sent on call being over the internet a dedicated service for voice
establishment. The “FastStart” element carries the calls is not possible unlike the old PSTN system.
proposed media channel description defining the This can lead to other factors affecting the quality of
media capabilities of the origin of the call allowing service as well as the limitations on ADSL upload
media communication to begin after one round-trip speed. This introduces many problems such as
message exchange instead of three. delay, packet loss, bandwidth limitations and echo.
With a normal ADSL connection (512:256) users are
5. COMPARISON BETWEEN H.323 AND likely to experience latencies of between 80ms and
SIP 400ms on a call, at around 200ms the flow of
conversation becomes distorted. This is mainly due
Both SIP and H.323 protocols can be used to most residential ADSL only supplying minimal
efficiently to connect client to client calls using any upload bandwidth thus limiting the user’s
range of media codecs. By placing a suitable proxy capabilities. All this is likely to change firstly when
server in the middle is it also possible to perform a SDSL becomes more widely available and then as
SIP to H.323 client call. Problems arise when BT themselves switch to IP based networks over the
complex functions such conference calls are next 5 years, known as 21CN [6][7].
attempted. H.323 is based on a centralized server
that uses a set of tightly integrated protocols to 7. VOIP SERVER PACKAGES
control sessions and users connections. In contrast,
SIP is often without a server, and its control There are many different VoIP server packages
mechanisms are much more loosely coupled and available each with their advantages over the other
depend a lot more on the client technology. SIP and each of their own complexity. At the time of
clients are able to join and leave a conference by writing Asterisk1 was the most popular server
using UDP signalling without the need for package providing built-in support for both H.323
centralised control. A central server in SIP is able to and SIP with functionality which can be harnessed
provide easier location of clients as well as after only about 20 minutes for setup. SIP Express
centralised session announcement (SAP) using the Router (SER)2 from iptel.org provides a proxy/router
Session Description Protocol (SDP). SIP provides a for SIP sessions with an optional extensions module
far more abstract protocol than H.323 able to be for the construction of a VoIP server. To set up a
used outside of systems such as VoIP, such as that PSTN like system in SER would take a lot longer
of multicast and unicast video streaming. H.323 is than the 20minutes of Asterisk, however SER
based around only a few recommendations and provides its own scripting language for complete
thus becomes easier to pick up and use for user control of functionality. Finally the last major
developers. A centralized architecture such as that server package is VOCAL3, a much more
of H.323 is more preferable to government services commercial solution with greater support for
which can still keep and eye on usage and enable a business and enterprise users. VOCAL is an open
line to be tapped, this being a requirement of any source project primarily designed as a SIP
ITU phone network. As SIP is not specifically softswitch, however includes translator plug-ins for
designed for telephony it does not have to comply support of H.323 endpoints. The following section
with this ruling as yet [8]. provides a more in depth look at the technology of
the 3 main server technologies and their
6. VOIP CODECS differences.
VoIP codecs are used to convert an analogue voice Being the most popular at the time of writing
signal into a digitally encoded version for Asterisk provides high levels of built in functionality
transmission over the internet. The same codec is which is easy to manage. Its ability to be able to
then used for the opposite purpose at its handle complex dialling plans and a wide range of
destination. Codecs vary greatly in sound quality, voice, fax, text and video codecs for direct
bandwidth required and computational interaction with users is able to seamlessly provide
requirements. Each server, program, gateway, etc switchboard, voicemail and operator support on the
typically supports several different codecs use of server. Asterisk supports both the H.323 and SIP
which use is negotiated upon initialization of the call. standards and most popular codecs, those used for
Server codec support is only required if the server is interaction with the user are shown in Figure 5.
able to interact with the client in operations such as Asterisk also provides the proprietary IAX (Inter-
switchboards and voicemail. Both SIP and H.323 Asterisk eXchange) protocol to enable the
contain abstraction layers supporting a set of 1 Asterisk – http://www.asterisk.org
standard codecs defined by the ITU for voice calls, 2 SER – http://www.iptel.org/ser
being more built into H.323 than SIP. Appendix A 3 VOCAL – http://www.vovida.org/
5. interconnection of multiple Asterisk servers, with the Codec Asterisk SER VOCAL
ability to forward communications between servers G. 711 Y Y
or use one server as backup for another. IAX G.723.1 Y
supplies a facility for VoIP with the same G.726 Y
functionality as an LEC within a PSTN’s LATA. G.729 Y
Asterisk provides a complete worldwide solution GSM 06.10 Y Y
such as that currently provided by the existing
LPC10e Y
PSTN network and is able to seamlessly interface
iLBC Y
with the PSTN network by making use of specifically
Speex Y
designed hardware. By deploying this hardware in
the correct fashion Asterisk is able to fulfil many of Figure 5 –Server support for codecs.
the ITU and FCC regulations i.e. enabling users to
be able to contact the emergency services from any Finally VOCAL from Vovida provides a much more
handset. This is achieved by adding a simple structured enterprise solution which is designed as
extension rule to the system to forward the a SIP softswitch however translators are available to
appropriate numbers onto the correct end users. allow interoperability with H.323 and MGCP
Asterisk provides both a structured number endpoints. The aim of the VOCAL project is to
identification as well as the newer provide a SIP based replacement to the PBX/PSTN
user@host.domain identification which is stored in a without necessarily providing any extra functionality
simple xml type extensions file which Asterisk reads such as that of Asterisk and SER. Vocal is designed
upon startup. Being designed to run on the UNIX to run on a distributed architecture of servers
platform Asterisk contains modules which can be providing redundancy to handle downtime and high
plugged in to enable user, extensions and calling volumes of usage [2]. Not providing any functionality
profiles to be managed by a web interface system such as voicemail at the server end means VOCAL
and harness the UNIX server capabilities. The one does not need any support for codecs, services
downside of the current implementation of Asterisk such as voicemail and call holding are left to the
(version 1.0) is the lack of support for IPv6 which individual clients rather than being supplied
would be required for large scale networks and for centrally. This makes VOCAL the much more
users currently behind a NAT (Network Address extensible system however from a users point a
Translation) or Firewall. view is not the easiest solution. Being an open
source project VOCAL also has to be installed from
SIP Express Router (SER) is a high performance, source as can both Asterisk and SER (which both
configurable, VoIP server supporting only SIP have more binary support however). VOCAL is a
clients and services. SER uses a full scripting very formidable package with the source code
language for its configuration, cutting down on the coming in at 78.1Mb, as opposed to 9.8Mb for SER
number of individual configuration files and or 37Mb for Asterisk.
improving scalability. This comes at the expense of
requiring operators to learn a new language and to 8. VOIP CLIENTS
mimic all the functionality which comes built into
Asterisk would require a very large learning curve. Clients and phones for use in VoIP networks are
To help Iptel the founders of SER provide many pre- now available as both hardware and software
built modules for plugging into SER including solutions; both being able to carry out the same
interface modules, accounting support and functionality of registering with a VoIP server to
voicemail. SERs primary intended use is as a SIP enable calls to be made to other clients. Wireless
proxy/router however also provides features to act handsets and mobile DECT technology phones are
as a registrar and redirect server. As a proxy/router also now available which are simply plugged into a
SER is designed to act as a standalone server and net cable at the base station rather than a phone
provides no functionality for direct communication cable. Although this technology is now becoming
with other servers. Redirection is built in, but this more widely available, users are still reluctant to buy
does not guarantee the user a connection and they into the hardware field as upgrade opportunities are
could just end up on a redirection loop or chain of limited without cost. Many solutions are at present
servers. Unlike Asterisk, SER has inbuilt support for reflecting the server technology they are designed
IPv6 as well as IPv4, and can listen for connections to connect to, the more advanced containing
on ports under both protocols concurrently. IPv6 answering machines and call holding ideal for use
capability provides greater support for mobile clients with a VOCAL server. Hardware phones are much
using the mobility headers of IPv6 and support for more commonly supporting the SIP protocol with
those users previously behind a NAT. SER provides H.323 support being relatively hard to find. This is
minimal extra interaction with users and support reflected in servers such as VOCAL where a
voicemail using a minimal set of base codecs as translations library for H.323 is provided as an add-
shown in Figure 5. on rather than a built-in.
6. Software solutions are much more variable in their as local deployment as a primary telephony service.
implementation and usage with upgrades being The project is looking at the many types of server
released at a constant rate to keep up with the technology and using both hardware and software
changing field and server technology. As with most clients to connect across a WAN with the additional
software many open source and free solutions can involvement of some willing volunteers from the
be found which install on any platform. Examples of Southampton Open Wireless Network (SOWN)6.
commercially produced clients include Windows Currently the system is operating on a single
Messenger1 for Windows and SJPhone2 for Asterisk server based in the main campus building
Windows and Linux. Neither of these handling all users and calling profiles. A SER server
implementations currently supports IPv6, and is also available on the same machine and is acting
Windows Messenger lacks DTMF (dial tone) as a proxy to Asterisk to provide IPv6 support for
generation facilities which prevent its use with testing with those clients which are able to use this
Voicemail and other touch-tone operated services. protocol. With SER acting as a proxy all extensions
Free and open source implementations include and user authentication is passed directly through to
KPhone3 and LinPhone4, both available for Linux. Asterisk by rewriting the incoming host port and
KPhone uses the KDE Qt library, while LinPhone translating the packets for use by Asterisk. This
has a GNOME GTK-based graphical interface. This solution operates without problems for calls placed
overview provides only a small selection from the between IPv4 users and IPv6 users providing both
ever expanding field where more companies are clients are using the same IP protocol. However if
getting involved on a daily basis. A recommended each client is using a different protocol the system
site to keep up with the latest is www.voip-info.org will fail on direct connection of the call due to the
which provides both listings of clients and server different connection types. To enable IPv4-IPv6
technology as well as useful guides to aid along the calls to be connected a further proxy has to be
way. provided to translate the packet headers to enable
each client to understand the data. This RTP-Proxy
As with the server technology the more popular
has to have support for both IPv4 and IPv6 traffic on
clients, software and hardware, mainly support the
the given network and the call between the clients
IPv4 protocol with limited support for IPv6. At the
should be routed through this proxy. Effectively an
time of writing IPv6 support in physical handsets
RTP-Proxy operates as a false client to which both
was still commercially unavailable. KPhone has
real clients send their information thinking that it is
been patched to support IPv6 using SER as the
their actual endpoint. The proxy then handles the
server, but this version has now become
traffic between the clients. This idea is good in
superseded and no longer works with most recent
prospect however research has shown not many of
release of SER and the Linux Kernel. LinPhone has
the client software packages currently support the
built in support for IPv6 and is being developed with
RTP-Proxy redirect information and still try to
this in mind, however this software implementation
connect the call directly [10]. This research is still on
is still at beta and contains many bugs [9].
going under the 6NET project being run worldwide
One of the most popular clients available currently including a section at the University.
operates on the Skype system which provides pure
VoIP using SIP (IPv4 only) through their own 10. CONCLUSION
software package. This has been designed with VoIP technology and distribution is on the increase
similar usage to MSN Messenger where a client is currently with many companies and institutes
able to have a phonebook of users who can be seen carrying out research in the area to further enhance
to be online or offline. This idea is now being this field, with the aim to provide the complete
expanded to include answering machine features solution. The two main protocols of SIP and H.323
and also to support various hardware phones now vary dramatically in their construction and usage of
becoming available with support for the Skype each has to be considered carefully. With the recent
system (www.skype.com). ruling by the FCC that SIP is not a specific
telephony protocol we now have a dramatic
9. VOIP IN OPERATION difference in the market between the two.
In this section a brief overview is offered as to the Governments are now backing the use of H.323 due
current state of research being performed at the to the legislation existing providing the ability to
University of Southampton into VoIP. This research monitor and control the networks usage. While
is being carried out in the Intelligence, Agents and 4 Windows Messenger -
Multimedia Group5 within the School of Electronics http://www.microsoft.com/windows/messenger/
and Computer Science and is focused on network 5 SJPhone – http://www.sjlabs.com
interoperability between IPv4 and IPv6 both over a 6 KPhone – http://www.iptel.org/products/kphone/
wired and wireless medium. The school regards 7 LinPhone – http://www.linphone.org
VoIP as an appropriate technology for roaming 8 IAM Research Group – http://www.iam.ecs.soton.ac.uk
academics (using mobility provided by IPv6) as well 9 SOWN – http://www.sown.org.uk
7. computing companies such as Cisco are backing REFERENCES
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8. 11. Appendix A
Codecs used in VoIP for communication between clients showing bit rate (quality) and bandwidth
consumption of each.
Standard bit rate sampling Raw Bandwidth
Name Description Remarks
by (kb/s) rate (kHz) Usage
(ADPCM
Intel, IMA ADPCM 32 8 var
) DVI
Also known as ulaw/alaw, mu-law
G.711 ITU-T Pulse code modulation (PCM) 64 8 87.2 Kbps
(US, Japan) and A-law (Europe)
Subband-codec that divides 16 kHz
G.722 ITU-T 7 kHz audio-coding within 64 kbit/s 64 16 * 120 Kbps + band into two subbands, each coded
using ADPCM
Coding at 24 and 32 kbit/s for hands-
G.722.1 ITU-T free operation in systems with low 24/32 16 * 60 Kbps + Variable Frame Size
frame loss
Dual rate speech coder for multimedia
Part of H.324 video conferencing.
G.723.1 ITU-T communications transmitting at 5.3 5.3/6.4 8 20.8/21.9 Kbps
DSP Group.
and 6.3 kbit/s
40, 32, 24, 16 kbit/s adaptive
16/24/32 31.5/47.2/55.2/6
G.726 ITU-T differential pulse code modulation 8 ADPCM; replaces G.721 and G.723.
/40 3.4 Kbps
(ADPCM)
5-, 4-, 3- and 2-bit/sample embedded
G.727 ITU-T adaptive differential pulse code var. ? var ADPCM. Related to G.726.
modulation (ADPCM)
Coding of speech at 16 kbit/s using
CELP. Annex J offers variable-bit
G.728 ITU-T low-delay code excited linear 16 8 31.5 Kbps
rate operation for DCME.
prediction
Coding of speech at 8 kbit/s using
G.729 ITU-T conjugate-structure algebraic-code- 8 8 31.2 Kbps Low delay (15 ms)
excited linear-prediction (CS-ACELP)
GSM Regular Pulse Excitation Long-Term
ETSI 13 8 30.3 Kbps Used for GSM cellular telephony.
06.10 Predictor (RPE-LTP)
10 coefficients. Also known as FIPS
LPC10e US Govt. Linear-predictive codec 2.4 8 7.8 Kbps
1015
iLBC (internet Low Bitrate Codec) Frames are encoded completely
iLBC IETF 13.3 8 27.7 Kbps
designed for narrow band speech. independently.
Speex is based on CELP and is
2.15-
Speex N/A designed to compress voice at bitrates 8/16/32 7.4 Kbps + open-source, multirate codec
44.2
ranging from 2 to 44 kbps.