Performance of VoIP with DCCP for satellite links

We present experimental results for the performance of selected voice codecs using the Datagram Congestion Control Protocol (DCCP) with TCP-Friendly Rate Control (TFRC) congestion control mechanism over a satellite link. We evaluate the performance of both constant and variable data rate speech codecs (G.729, G.711 and Speex) for a number of simultaneous calls, using the ITU E-model and identify problem areas and potential for improvement. Our experiments are done on a commercial satellite service using a data stream generated by a VoIP application, configured with selected voice codecs and using the DCCP/CCID4 Linux implementation. We analyse the sources of packet losses which are a main contributor to reduced voice quality when using CCID4 and additionally analyse the effect of jitter which is one of the crucial parameters contributing to VoIP quality and has, to the best of our knowledge, not been considered previously in the published DCCP performance results. We propose modifications to the CCID4 algorithm and demonstrate how these improve the VoIP performance, without the need for additional link information other than what is already monitored by CCID4 (which is the case for Quick-Start). We also demonstrate the fairness of the proposed modifications to other flows. We identify the additional benefit of DCCP when used in VoIP admission control mechanisms and draw conclusions about the advantages and disadvantages of the proposed DCCP/ CCID4 congestion control mechanism for use with VoIP applications

Retooling for Success: A Case Study of VoIP Implementation to Improve Customer Service at a Midwestern Financial Services Office

This article presents a case study of the acquisition of a Voice over Internet Protocol (VoIP) system to replace an outdated telephone system at a Midwestern financial services company. In the Midwest retail office of ABC Financial Services, the old Private Branch Exchange (PBX) phone system was incapable of handling customer inquiries during the busy tax return season. The inefficient systems exposed the organization to missed and delayed calls, which lead to a considerable number of customer complaints and lost revenue. Guided by the systematic approach of technology retooling, computer engineers followed the steps of problem diagnosis, analysis of competing solutions, implementation, and assessment of the VoIP system as the replacement telecommunications platform. System performance and evaluation data were collected during and after system implementation. Assessment of the new VoIP system demonstrated improved availability, speed, and reliability of the information provided to customers. New functionalities, such as customer inquiry of the database, provided through the VoIP system pushed the self-service adoption to a record high level. The system implementation also fosters an updated IT plan that will help this organization chart its business strategy for future years

Design and implementation of a Marking Strategy to Increase the Contactability in the Call Centers, Based on Machine Learning

Jamar is a company that belongs to the furniture sector, which manufactures and sells furniture and accessories for the home. Customer calls are one of the most trusted channels used in contact centers. Currently, the contactability indicator has a goal of 40% and is at 31%. The enemies of the efficiency of this channel are the terrible dimensioning, customers who evade answering these calls by identifying the numbers, non-market numbers in the databases, failures in the technological resources. Therefore, a proposal was made to design and implement a marking strategy in the call center, supported by a statistical model for dimensioning. Likewise, emerging technology such as Machine Learning is performed to help the marking strategy in outbound campaigns, reconfiguration of the dialplan to make it more efficient, and a redundant architecture design in the operators. Basic concepts of Teletraffic are explained, showing its primary functions, relevant for the management of the company's telephone system. In the same way, fundamentals of the Asterisk IP PBX are exposed, one of the most used in our environment due to its versatility and low implementation cost. The methodology of descriptive and applied research is used for the development of the project. The results and discussion show the dialing strategy and some call statistics from previous years, necessary to establish a correct dimensioning of the solution. The proposed solution allows having redundancy management for SIP and trunk operators, to have backup and reliability in case of failure

A Proposal for A High Availability Architecture for VoIP Telephone Systems based on Open Source Software

The inherent needs of organizations to improve and amplify their technological platform entail large expenses with the goal to enhance their performance. Hence, they have to contemplate mechanisms of optimization and the improvement of their operational infrastructure. In this direction arises the need to guarantee the correct operation and non-degradation of the services provided by the platform during the periods with a significant load of work. This type of scenario is perfectly applicable to the field of VoIP technologies, where users generate elevated loads of work on critical points of the infrastructure, during the process of interaction with their peers. In this research work, we propose a solution for high availability, with the goal of maintaining the continuity of the operation of communication environments based on the SIP protocol in high load. We validate our proposal through numerous experiments. Also, we compare our solution with other classical VoIP scenarios and show the advantages of a high availability and fault tolerance architecture for organizations

On the quality of VoIP with DCCP for satellite communications

We present experimental results for the performance of selected voice codecs using DCCP with CCID4 congestion control over a satellite link. We evaluate the performance of both constant and variable data rate speech codecs for a number of simultaneous calls using the ITU E-model. We analyse the sources of packet losses and additionally analyse the effect of jitter which is one of the crucial parameters contributing to VoIP quality and has, to the best of our knowledge, not been considered previously in the published DCCP performance results. We propose modifications to the CCID4 algorithm and demonstrate how these improve the VoIP performance, without the need for additional link information other than what is already monitored by CCID4. We also demonstrate the fairness of the proposed modifications to other flows. Although the recently adopted changes to TFRC specification alleviate some of the performance issues for VoIP on satellite links, we argue that the characteristics of commercial satellite links necessitate consideration of further improvements. We identify the additional benefit of DCCP when used in VoIP admission control mechanisms and draw conclusions about the advantages and disadvantages of the proposed DCCP/CCID4 congestion control mechanism for use with VoIP applications

Design and Implementation VOIP Service on Open IMS and Asterisk Servers Interconnected through Enum Server

Asterisk and Open IMS use SIP signal protocol to enable both of them can be connected. To facilitate both relationships, Enum server- that is able to translate the numbering address such as PSTN (E.164) to URI address (Uniform Resource Identifier)- can be used. In this research, we interconnect Open IMS and Asterisk server Enum server. We then analyze the server performance and PDD (Post Dial Delay) values resulted by the system. As the result of the experiment, we found that, for a call from Open IMS user to analog Asterisk telephone (FXS) with a arrival call each servers is 30 call/sec, the maximum PDD value is 493.656 ms. Open IMS is able to serve maximum 30 call/s with computer processor 1.55 GHz, while the Asterisk with computer processor 3.0 GHz, may serve up to 55 call/sec. Enum on server with 1.15 GHz computer processor have the capability of serving maximum of 8156 queries/sec.Comment: 12 Page

Evaluation of Voip Technologies As a Replacement for Traditional Pstn Based Pbx Systems

This project deals with a company in the SME sector with offices located in the midlands of Ireland. The company is well established in the field of Agri-Feed Manufacturing Process Control Systems or SCADA systems, and has been established for over 20 years. The communications requirements of the company have changed over these 20 plus years to a mix of various technologies from PSTN lines to Broadband ADSL. The present telephone system has been in use since 1991 and has several questions marks over it in terms of usage costs, usage reporting, support and maintenance and features available. This project is an evaluation of the possible benefits offered by the use of VOIP technologies and Asterisk Open Source PBX as a possible replacement for the existing telephone system in place. It attempts to look at the potential benefits costs, and risks associated with using such a system. A small pilot system is implemented and some key users test this and feedback on its usability is recorded. The current communications infrastructure is analysed in an effort to highlight systems where cost savings or benefits can be made by switching to these other technologies and a report was presented to the management in order to give the required information to make the best possible informed decision about the way forward for the company

Aplicabilidad de telefonía IP en la computación en la nube

This paper carries out a research related to the applicability of VoIP over Cloud Computing to guarantee service stability and elasticity of the organizations. In this paper, Elastix is used as an open source software that allows the management and control of a Private Branch Exchange (PBX); and for developing, it is used the services given Amazon Web Services due to their leadership and experience in cloud computing providing security, scalability, backup service and feasibility for the users.Este trabajo lleva a cabo una investigación relacionada con la aplicabilidad de VoIP sobre Cloud Computing para garantizar la estabilidad del servicio y la elasticidad de las organizaciones. En este documento, Elastix se utiliza como un software de código abierto que permite gestión y control de una central telefónica privada (PBX); y para el desarrollo, se utilizan los servicios prestados a Amazon Web Services debido a su liderazgo y experiencia en computación en la nube que brinda seguridad, escalabilidad y servicio de respaldo y viabilidad para los usuarios