A revenue generation model for adoption of voice telephony applications
- Authors: Mazwi, Sekelo P Lusanda
- Date: 2015-01
- Subjects: Internet telephony , Telecommunication systems
- Language: English
- Type: text
- Identifier: http://hdl.handle.net/10353/25530 , vital:64331
- Description: Amongst various Voice Telephony Applications discovered, IBM India has developed the Spoken Web which enables voice commerce capabilities that are ideally suitable for underprivileged rural communities. Admittedly, these communities are barred from fully utilising ICT enabled electronic commerce services such as the Internet for business marketing processes as a consequence of poor literacy and financial constraints. The advent of Voice Telephony Applications aids accessibility and participation of the underprivileged rural communities to the ICT world. It should be noted that users are not compelled to be computer savvy to make voice calls, but high voice call costs are the deterrent. Therefore, devising an appropriate Revenue Generation Model would enhance the accessibility and participation of the underprivileged rural communities to the ICT world. As guided by the Design Science Approach, and the Diffusion of Innovations and Social Exchange theories, this research project has discovered and applied five models such as the Freemium, Affiliation, Advertising, Incentives-driven and Subsidy within each development stage of Spoken Web. This research project is of the idea that reimbursing content providers through supporting the proposed Revenue Generation Model as a de facto solution will help in reducing voice call costs to the users of Voice Telephony Applications in future. , Thesis (MCom) -- Faculty of Management and Commerce, 2015
- Full Text:
- Date Issued: 2015-01
- Authors: Mazwi, Sekelo P Lusanda
- Date: 2015-01
- Subjects: Internet telephony , Telecommunication systems
- Language: English
- Type: text
- Identifier: http://hdl.handle.net/10353/25530 , vital:64331
- Description: Amongst various Voice Telephony Applications discovered, IBM India has developed the Spoken Web which enables voice commerce capabilities that are ideally suitable for underprivileged rural communities. Admittedly, these communities are barred from fully utilising ICT enabled electronic commerce services such as the Internet for business marketing processes as a consequence of poor literacy and financial constraints. The advent of Voice Telephony Applications aids accessibility and participation of the underprivileged rural communities to the ICT world. It should be noted that users are not compelled to be computer savvy to make voice calls, but high voice call costs are the deterrent. Therefore, devising an appropriate Revenue Generation Model would enhance the accessibility and participation of the underprivileged rural communities to the ICT world. As guided by the Design Science Approach, and the Diffusion of Innovations and Social Exchange theories, this research project has discovered and applied five models such as the Freemium, Affiliation, Advertising, Incentives-driven and Subsidy within each development stage of Spoken Web. This research project is of the idea that reimbursing content providers through supporting the proposed Revenue Generation Model as a de facto solution will help in reducing voice call costs to the users of Voice Telephony Applications in future. , Thesis (MCom) -- Faculty of Management and Commerce, 2015
- Full Text:
- Date Issued: 2015-01
Service provisioning in two open-source SIP implementation, cinema and vocal
- Authors: Hsieh, Ming Chih
- Date: 2013-06-18
- Subjects: Real-time data processing , Computer network protocols , Internet telephony , Digital telephone systems , Communication -- Technological innovations
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4687 , http://hdl.handle.net/10962/d1008195 , Real-time data processing , Computer network protocols , Internet telephony , Digital telephone systems , Communication -- Technological innovations
- Description: The distribution of real-time multimedia streams is seen nowadays as the next step forward for the Internet. One of the most obvious uses of such streams is to support telephony over the Internet, replacing and improving traditional telephony. This thesis investigates the development and deployment of services in two Internet telephony environments, namely CINEMA (Columbia InterNet Extensible Multimedia Architecture) and VOCAL (Vovida Open Communication Application Library), both based on the Session Initiation Protocol (SIP) and open-sourced. A classification of services is proposed, which divides services into two large groups: basic and advanced services. Basic services are services such as making point-to-point calls, registering with the server and making calls via the server. Any other service is considered an advanced service. Advanced services are defined by four categories: Call Related, Interactive, Internetworking and Hybrid. New services were implemented for the Call Related, Interactive and Internetworking categories. First, features involving call blocking, call screening and missed calls were implemented in the two environments in order to investigate Call-related services. Next, a notification feature was implemented in both environments in order to investigate Interactive services. Finally, a translator between MGCP and SIP was developed to investigate an Internetworking service in the VOCAL environment. The practical implementation of the new features just described was used to answer questions about the location of the services, as well as the level of required expertise and the ease or difficulty experienced in creating services in each of the two environments. , KMBT_363 , Adobe Acrobat 9.54 Paper Capture Plug-in
- Full Text:
- Authors: Hsieh, Ming Chih
- Date: 2013-06-18
- Subjects: Real-time data processing , Computer network protocols , Internet telephony , Digital telephone systems , Communication -- Technological innovations
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4687 , http://hdl.handle.net/10962/d1008195 , Real-time data processing , Computer network protocols , Internet telephony , Digital telephone systems , Communication -- Technological innovations
- Description: The distribution of real-time multimedia streams is seen nowadays as the next step forward for the Internet. One of the most obvious uses of such streams is to support telephony over the Internet, replacing and improving traditional telephony. This thesis investigates the development and deployment of services in two Internet telephony environments, namely CINEMA (Columbia InterNet Extensible Multimedia Architecture) and VOCAL (Vovida Open Communication Application Library), both based on the Session Initiation Protocol (SIP) and open-sourced. A classification of services is proposed, which divides services into two large groups: basic and advanced services. Basic services are services such as making point-to-point calls, registering with the server and making calls via the server. Any other service is considered an advanced service. Advanced services are defined by four categories: Call Related, Interactive, Internetworking and Hybrid. New services were implemented for the Call Related, Interactive and Internetworking categories. First, features involving call blocking, call screening and missed calls were implemented in the two environments in order to investigate Call-related services. Next, a notification feature was implemented in both environments in order to investigate Interactive services. Finally, a translator between MGCP and SIP was developed to investigate an Internetworking service in the VOCAL environment. The practical implementation of the new features just described was used to answer questions about the location of the services, as well as the level of required expertise and the ease or difficulty experienced in creating services in each of the two environments. , KMBT_363 , Adobe Acrobat 9.54 Paper Capture Plug-in
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Constructing a low-cost, open-source, VoiceXML
- Authors: King, Adam
- Date: 2007 , 2013-07-01
- Subjects: VoiceXML (Document markup language) , Asterisk (Computer file) , Internet telephony , Open source software
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4585 , http://hdl.handle.net/10962/d1004735 , VoiceXML (Document markup language) , Asterisk (Computer file) , Internet telephony , Open source software
- Description: Voice-enabled applications, applications that interact with a user via an audio channel, are used extensively today. Their use is growing as speech related technologies improve, as speech is one of the most natural methods of interaction. They can provide customer support as IVRs, can be used as an assistive technology, or can become an aural interface to the Internet. Given that the telephone is used extensively throughout the globe, the number of potential users of voice-enabled applications is very high. VoiceXML is a popular, open, high-level, standard means of creating voice-enabled applications which was designed to bring the benefits of web based development to services. While VoiceXML is an ideal language for creating these applications, VoiceXML gateways, the hardware and software responsible for interpreting VoiceXML applications and interfacing with the PSTN, are still expensive and so there is a need for a low-cost gateway. Asterisk, and open-source, TDM/VoIP telephony platform, can be used as a low-cost PSTN interface. This thesis investigates adding a VoiceXML service to Asterisk, creating a low-cost VoiceXML prototype gateway which is able to render voice-enabled applications. Following the Component-Based Software Engineering (CBSE) paradigm, the VoiceXML gateway is divided into a set of components which are sourced from the open-source community, and integrated to create the gateway. The browser requires a VoiceXML interpreter (OpenVXI), a Text-To-Speech engine (Festival) and a speech recognition engine (Sphinx 4). The integration of the components results in a low-cost, open-source VoiceXML gateway. System tests show that the integration of the components was successful, and that the system can handle concurrent calls. A fully compliant version of the gateway can be used in the real world to render voice-enabled applications at a low cost. , KMBT_363 , Adobe Acrobat 9.55 Paper Capture Plug-in
- Full Text:
- Date Issued: 2007
- Authors: King, Adam
- Date: 2007 , 2013-07-01
- Subjects: VoiceXML (Document markup language) , Asterisk (Computer file) , Internet telephony , Open source software
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4585 , http://hdl.handle.net/10962/d1004735 , VoiceXML (Document markup language) , Asterisk (Computer file) , Internet telephony , Open source software
- Description: Voice-enabled applications, applications that interact with a user via an audio channel, are used extensively today. Their use is growing as speech related technologies improve, as speech is one of the most natural methods of interaction. They can provide customer support as IVRs, can be used as an assistive technology, or can become an aural interface to the Internet. Given that the telephone is used extensively throughout the globe, the number of potential users of voice-enabled applications is very high. VoiceXML is a popular, open, high-level, standard means of creating voice-enabled applications which was designed to bring the benefits of web based development to services. While VoiceXML is an ideal language for creating these applications, VoiceXML gateways, the hardware and software responsible for interpreting VoiceXML applications and interfacing with the PSTN, are still expensive and so there is a need for a low-cost gateway. Asterisk, and open-source, TDM/VoIP telephony platform, can be used as a low-cost PSTN interface. This thesis investigates adding a VoiceXML service to Asterisk, creating a low-cost VoiceXML prototype gateway which is able to render voice-enabled applications. Following the Component-Based Software Engineering (CBSE) paradigm, the VoiceXML gateway is divided into a set of components which are sourced from the open-source community, and integrated to create the gateway. The browser requires a VoiceXML interpreter (OpenVXI), a Text-To-Speech engine (Festival) and a speech recognition engine (Sphinx 4). The integration of the components results in a low-cost, open-source VoiceXML gateway. System tests show that the integration of the components was successful, and that the system can handle concurrent calls. A fully compliant version of the gateway can be used in the real world to render voice-enabled applications at a low cost. , KMBT_363 , Adobe Acrobat 9.55 Paper Capture Plug-in
- Full Text:
- Date Issued: 2007
Decorating Asterisk : experiments in service creation for a multi-protocol telephony environment using open source tools
- Authors: Hitchcock, Jonathan
- Date: 2006
- Subjects: Asterisk (Computer file) , Internet telephony
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4635 , http://hdl.handle.net/10962/d1006539 , Asterisk (Computer file) , Internet telephony
- Description: As Voice over IP becomes more prevalent, value-adds to the service will become ubiquitous. Voice over IP (VoIP) is no longer a single service application, but an array of marketable services of increasing depth, which are moving into the non-desktop market. In addition, as the range of devices being generally used increases, it will become necessary for all services, including VoIP services, to be accessible from multiple platforms and through varied interfaces. With the recent introduction and growth of the open source software PBX system named Asterisk, the possibility of achieving these goals has become more concrete. In addition to Asterisk, a number of open source systems are being developed which facilitate the development of systems that interoperate over a wide variety of platforms and through multiple interfaces. This thesis investigates Asterisk in terms of its viability to provide the depth of services that will be required in a VoIP environment, as well as a number of other open source systems in terms of what they can offer such a system. In addition, it investigates whether these services can be made available on different devices. Using various systems built as a proof-of-concept, this thesis shows that Asterisk, in conjunction with various other open source projects, such as the Twisted framework provides a concrete tool which can be used to realise flexible and protocol independent telephony solutions for a small to medium enterprise.
- Full Text:
- Date Issued: 2006
- Authors: Hitchcock, Jonathan
- Date: 2006
- Subjects: Asterisk (Computer file) , Internet telephony
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4635 , http://hdl.handle.net/10962/d1006539 , Asterisk (Computer file) , Internet telephony
- Description: As Voice over IP becomes more prevalent, value-adds to the service will become ubiquitous. Voice over IP (VoIP) is no longer a single service application, but an array of marketable services of increasing depth, which are moving into the non-desktop market. In addition, as the range of devices being generally used increases, it will become necessary for all services, including VoIP services, to be accessible from multiple platforms and through varied interfaces. With the recent introduction and growth of the open source software PBX system named Asterisk, the possibility of achieving these goals has become more concrete. In addition to Asterisk, a number of open source systems are being developed which facilitate the development of systems that interoperate over a wide variety of platforms and through multiple interfaces. This thesis investigates Asterisk in terms of its viability to provide the depth of services that will be required in a VoIP environment, as well as a number of other open source systems in terms of what they can offer such a system. In addition, it investigates whether these services can be made available on different devices. Using various systems built as a proof-of-concept, this thesis shows that Asterisk, in conjunction with various other open source projects, such as the Twisted framework provides a concrete tool which can be used to realise flexible and protocol independent telephony solutions for a small to medium enterprise.
- Full Text:
- Date Issued: 2006
Investigating call control using MGCP in conjuction with SIP and H.323
- Authors: Jacobs, Ashley
- Date: 2005 , 2005-03-14
- Subjects: Communication -- Technological innovations , Digital telephone systems , Computer networks , Computer network protocols , Internet telephony
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4631 , http://hdl.handle.net/10962/d1006516 , Communication -- Technological innovations , Digital telephone systems , Computer networks , Computer network protocols , Internet telephony
- Description: Telephony used to mean using a telephone to call another telephone on the Public Switched Telephone Network (PSTN), and data networks were used purely to allow computers to communicate. However, with the advent of the Internet, telephony services have been extended to run on data networks. Telephone calls within the IP network are known as Voice over IP. These calls are carried by a number of protocols, with the most popular ones currently being Session Initiation Protocol (SIP) and H.323. Calls can be made from the IP network to the PSTN and vice versa through the use of a gateway. The gateway translates the packets from the IP network to circuits on the PSTN and vice versa to facilitate calls between the two networks. Gateways have evolved and are now split into two entities using the master/slave architecture. The master is an intelligent Media Gateway Controller (MGC) that handles the call control and signalling. The slave is a "dumb" Media Gateway (MG) that handles the translation of the media. The current gateway control protocols in use are Megaco/H.248, MGCP and Skinny. These protocols have proved themselves on the edge of the network. Furthermore, since they communicate with the call signalling VoIP protocols as well as the PSTN, they have to be the lingua franca between the two networks. Within the VoIP network, the numbers of call signalling protocols make it difficult to communicate with each other and to create services. This research investigates the use of Gateway Control Protocols as the lowest common denominator between the call signalling protocols SIP and H.323. More specifically, it uses MGCP to investigate service creation. It also considers the use of MGCP as a protocol translator between SIP and H.323. A service was created using MGCP to allow H.323 endpoints to send Short Message Service (SMS) messages. This service was then extended with minimal effort to SIP endpoints. This service investigated MGCP’s ability to handle call control from the H.323 and SIP endpoints. An MGC was then successfully used to perform as a protocol translator between SIP and H.323.
- Full Text:
- Date Issued: 2005
- Authors: Jacobs, Ashley
- Date: 2005 , 2005-03-14
- Subjects: Communication -- Technological innovations , Digital telephone systems , Computer networks , Computer network protocols , Internet telephony
- Language: English
- Type: Thesis , Masters , MSc
- Identifier: vital:4631 , http://hdl.handle.net/10962/d1006516 , Communication -- Technological innovations , Digital telephone systems , Computer networks , Computer network protocols , Internet telephony
- Description: Telephony used to mean using a telephone to call another telephone on the Public Switched Telephone Network (PSTN), and data networks were used purely to allow computers to communicate. However, with the advent of the Internet, telephony services have been extended to run on data networks. Telephone calls within the IP network are known as Voice over IP. These calls are carried by a number of protocols, with the most popular ones currently being Session Initiation Protocol (SIP) and H.323. Calls can be made from the IP network to the PSTN and vice versa through the use of a gateway. The gateway translates the packets from the IP network to circuits on the PSTN and vice versa to facilitate calls between the two networks. Gateways have evolved and are now split into two entities using the master/slave architecture. The master is an intelligent Media Gateway Controller (MGC) that handles the call control and signalling. The slave is a "dumb" Media Gateway (MG) that handles the translation of the media. The current gateway control protocols in use are Megaco/H.248, MGCP and Skinny. These protocols have proved themselves on the edge of the network. Furthermore, since they communicate with the call signalling VoIP protocols as well as the PSTN, they have to be the lingua franca between the two networks. Within the VoIP network, the numbers of call signalling protocols make it difficult to communicate with each other and to create services. This research investigates the use of Gateway Control Protocols as the lowest common denominator between the call signalling protocols SIP and H.323. More specifically, it uses MGCP to investigate service creation. It also considers the use of MGCP as a protocol translator between SIP and H.323. A service was created using MGCP to allow H.323 endpoints to send Short Message Service (SMS) messages. This service was then extended with minimal effort to SIP endpoints. This service investigated MGCP’s ability to handle call control from the H.323 and SIP endpoints. An MGC was then successfully used to perform as a protocol translator between SIP and H.323.
- Full Text:
- Date Issued: 2005
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