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
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- 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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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.
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- 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.
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