18-10-2010, 10:14 AM
This article is presented by:
Andrew Quinn
ABSTRACT
Internet Protocol (IP) Telephony has many issues that have to be overcome before it can be considered a rival to the existing telephony infrastructure. One such issue is the quality of service. The use of play-out buffering at the receiver helps to improve the quality of Voice over IP (VoIP). A buffering algorithm has been proposed by Narbutt, which uses a dynamic approach to buffering. This algorithm is adjusted automatically according to an estimate of the network delay. This is more suitable to the changing network conditions usually experienced. The buffer has been implemented using the H.323 signalling protocol.
The aim of this project is to incorporate Narbutt’s adaptive buffering algorithm into the Session Initiation Protocol (SIP). SIP has been shown to be much easier to implement and update than H.323. The integration of the algorithm was doing using VOCAL, a VoIP software library based around SIP. This report describes IP telephony and the protocols surrounding it, and the software used is also described. The manipulation of VoIP software to implement the play-out buffer and the issues involved in doing this are discussed.
INTRODUCTION
The ability to communicate properly over long distances has become an integral part of society today. Businesses are expanding to different regions in the world, but need to keep the same deadlines. This means it is necessary for employees in two different regions to communicate with each other over long distances, cheaply and trouble free. The public switched telephone network (PSTN) has developed itself to accommodate these requirements.
The internet has become a very popular means of communication in a very short period of time. It was set up as a network where people could share files and access other peoples work. It has since established itself as a massive communications infrastructure that provides many services such as electronic mail. In the recent years it has further developed itself into providing Internet Telephony or Voice over Internet Protocol (VoIP). This allows users to make voice or video calls over the internet. All the user needs is a computer with a network connection, a soundcard, and a microphone. VoIP enables a lot of big companies to combine their communications and their networking infrastructures. This is the biggest advantage that VoIP has over the regular telephone system. It means that voice and multimedia services are joined together. This means that a number of calls can be made on the one line, as well as having a multimedia broadcast. The fact that you are putting elements that would use one line each, down a single line, means that costs are significantly cut in the management and leasing of lines. There is even no need to change the communication infrastructure that already exists in the company. The companies PABX (private automatic branch exchange) only has to connect to a VoIP gateway, so IP calls can be made. Although VoIP seems to be taking off more with the corporate market the emergence and interest of the general public with Broadband should mean that IP telephony service could soon be implemented to its full extent in the home environment.
For more information about this article,please follow the link:
http://www.googleurl?sa=t&source=web&cd=...Report.pdf&ei=Rs67TNeOKceVcZ3-mfYM&usg=AFQjCNGmjJIOzE3QjXbfGliImJDisXzZKQ
Andrew Quinn
Implementing SIP for VoIP Algorithms
JM-3
JM-3
ABSTRACT
Internet Protocol (IP) Telephony has many issues that have to be overcome before it can be considered a rival to the existing telephony infrastructure. One such issue is the quality of service. The use of play-out buffering at the receiver helps to improve the quality of Voice over IP (VoIP). A buffering algorithm has been proposed by Narbutt, which uses a dynamic approach to buffering. This algorithm is adjusted automatically according to an estimate of the network delay. This is more suitable to the changing network conditions usually experienced. The buffer has been implemented using the H.323 signalling protocol.
The aim of this project is to incorporate Narbutt’s adaptive buffering algorithm into the Session Initiation Protocol (SIP). SIP has been shown to be much easier to implement and update than H.323. The integration of the algorithm was doing using VOCAL, a VoIP software library based around SIP. This report describes IP telephony and the protocols surrounding it, and the software used is also described. The manipulation of VoIP software to implement the play-out buffer and the issues involved in doing this are discussed.
INTRODUCTION
The ability to communicate properly over long distances has become an integral part of society today. Businesses are expanding to different regions in the world, but need to keep the same deadlines. This means it is necessary for employees in two different regions to communicate with each other over long distances, cheaply and trouble free. The public switched telephone network (PSTN) has developed itself to accommodate these requirements.
The internet has become a very popular means of communication in a very short period of time. It was set up as a network where people could share files and access other peoples work. It has since established itself as a massive communications infrastructure that provides many services such as electronic mail. In the recent years it has further developed itself into providing Internet Telephony or Voice over Internet Protocol (VoIP). This allows users to make voice or video calls over the internet. All the user needs is a computer with a network connection, a soundcard, and a microphone. VoIP enables a lot of big companies to combine their communications and their networking infrastructures. This is the biggest advantage that VoIP has over the regular telephone system. It means that voice and multimedia services are joined together. This means that a number of calls can be made on the one line, as well as having a multimedia broadcast. The fact that you are putting elements that would use one line each, down a single line, means that costs are significantly cut in the management and leasing of lines. There is even no need to change the communication infrastructure that already exists in the company. The companies PABX (private automatic branch exchange) only has to connect to a VoIP gateway, so IP calls can be made. Although VoIP seems to be taking off more with the corporate market the emergence and interest of the general public with Broadband should mean that IP telephony service could soon be implemented to its full extent in the home environment.
For more information about this article,please follow the link:
http://www.googleurl?sa=t&source=web&cd=...Report.pdf&ei=Rs67TNeOKceVcZ3-mfYM&usg=AFQjCNGmjJIOzE3QjXbfGliImJDisXzZKQ