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

Design, implementation and evaluation of unified communications on-premises and over the Cloud

Unified Communication (UC) is the integration of two or more real time communication systems into one platform. Integrating core communication systems into one overall enterprise level system delivers more than just cost saving. These real-time interactive communication services and applications over Internet Protocol (IP) have become critical in boosting employee accessibility and efficiency, improving customer support and fostering business agility. However, some small and medium-sized businesses (SMBs) are far from implementing this solution due to the high cost of initial deployment and ongoing support. Cloud based UC solution, UC as a Service (UCaaS), is now itself a maturing technology in the marketplace and it has revolutionized the IT industry, being the powerful platform that many businesses are choosing to migrate their on-premises UC solution onto. UCaaS solution has the potential to reduce the capital and operational expenses associated with deploying UC on their own. In this paper, we will discuss and demonstrate an open source on-premises UC solution, viz. “Asterisk” for use by businesses, and report on some performance tests using SIPp. This paper also discusses and demonstrates an open source UCaaS solution. The contribution from this research is the provision of technical advice to businesses in deploying UC and UCaaS, which is manageable in terms of cost, ease of deployment and support

Softswitch Design and Performance Analysis

The increasing number of subscribers’ demands in telecommunication sector has motivated the operators to provide high quality of service in cost effective way. Moreover, operators need to have an open structure system so that they can move their systems to the next generation network architecture. For this purpose, Softswitch is an appropriate technology because it is a safe and cost efficient solution and though it can migrate from traditional circuit-switching based telephone system to internet protocol packet-switching based networking. Softswitch network divides the logical switch into several parts with different functions such as signaling gateway, media gateway, media server, etc. Standard communication protocols are implemented between those parts. Softswitch is software-based system to make connection between devices, and moreover to control voice calls, data and routes calls through different entities of the networks. Softswitch supports management functions such as provisioning, fault handling and reporting, billing, operational support, etc. Softswitch suitable for all types of traffic and services so it is very demanding in the competitive world of mobile operators. In this thesis, Softswitch has been studied and analyzed in details. Softswitch network consisted of different integrated modules such as transportation, calling and signaling, service application and management. Each module provides different services such as call control, routing, billing and network management. Each module is discussed from functional and service point of views. Softswitch based wireless network architecture as well as variety of service solutions is presented. Different protocol interfaces in softswitch network such as signaling system number 7 are explained. Moreover, bearer calls, independent call control protocol, gateway control protocol, IP bearer control protocol are explained as well. Variety of softswitch network architectures analysis has been done based on their performance and the applicability. Three Softswitch network architectures are proposed which are validated through simulations.fi=Opinnäytetyö kokotekstinä PDF-muodossa.|en=Thesis fulltext in PDF format.|sv=Lärdomsprov tillgängligt som fulltext i PDF-format

A comparative study of in-band and out-of-band VOIP protocols in layer 3 and layer 2.5 environments

For more than a century the classic circuit-switched telephony in the form of PSTN (Public Service Telephone Network) has dominated the world of phone communications (Varshney et al., 2002). The alternative solution of VoIP (Voice over Internet Protocol) or Internet telephony has increased dramatically its share over the years though. Originally started among computer enthusiasts, nowadays it has become a huge research area in both the academic community as well as the industry (Karapantazis and Pavlidou, 2009). Therefore, many VoIP technologies have emerged in order to offer telephony services. However, the performance of these VoIP technologies is a key issue for the sound quality that the end-users receive. When making reference to sound quality PSTN still stands as the benchmark.Against this background, the aim of this project is to evaluate different VoIP signalling protocols in terms of their key performance metrics and the impact of security and packet transport mechanisms on them. In order to reach this aim in-band and out-of-band VoIP signalling protocols are reviewed along with the existing security techniques which protect phone calls and network protocols that relay voice over packet-switched systems. In addition, the various methods and tools that are used in order to carry out performance measurements are examined together with the open source Asterisk VoIP platform. The findings of the literature review are then used in order to design and implement a novel experimental framework which is employed for the evaluation of the in-band and out-of-band VoIP signalling protocols in respect to their key performance networks. The major issue of this framework though is the lack of fine-grained clock synchronisation which is required in order to achieve ultra precise measurements. However, valid results are still extracted. These results show that in-band signalling protocols are highly optimised for VoIP telephony and outperform out-of-band signalling protocols in certain key areas. Furthermore, the use of VoIP specific security mechanisms introduces just a minor overhead whereas the use of Layer 2.5 protocols against the Layer 3 routing protocols does not improve the performance of the VoIP signalling protocols

Transmission média sur les réseaux IP en utilisant les protocoles SIP et IAX

Les progrès technologiques du réseau Internet ont permis le développement de nouvelles applications multimédia; la voix, la vidéo et la vidéoconférence sont devenues des domaines importants de recherche et de développement pour l’industrie des télécommunications. Ces dernières années ont été remarquables par la mise en oeuvre de connexion haute débit, et de terminaux mobile et fixe performants. Plusieurs standards ont été conçus spécifiquement pour permettre la transmission média sur les réseaux IP avec une meilleure qualité de service. Ce travail a pour but d’étudier les protocoles de transmission média sur les réseaux IP, en commençant par l’état de l’art de technologies principales pour accéder au réseau, les techniques utilisées pour encoder l’audio et la vidéo, et en finissant par les protocoles de transport combinés avec d’autres protocoles temps réels. L’objectif principal du mémoire est d’analyser, et intégrer les protocoles de transmission (SIP, RTP et IAX) sur les réseaux IP. Le projet se compose de deux parties : expérimentale et applicative. La première partie a pour objectif de mettre en place une plateforme IPPBX capable de fournir une solution assez complète de transmission média sur le réseau IP en utilisant les protocoles SIP et IAX. Ensuite, nous allons calculer le temps requis de signalisation SIP/IAX et la qualité de service d’une communication IAX en utilisant les codecs G.711 et GSM. La deuxième partie se compose de la conception et l’implémentation du protocole RTP dans les téléphones mobiles en utilisant la technologie J2ME pour permettre un environnement mobile de vidéoconférence. Nous allons effectuer un rapport technique assez complet décrivant la technologie mobile J2ME. Nous allons également tester les émulateurs et outils capables d’offrir un environnement de vidéoconférence mobile et les difficultés associées aux codecs supportés. Les résultats des expériences ont montré que le temps requis de signalisation SIP et IAX est sous un seuil acceptable dans un réseau local. Selon les valeurs obtenues du délai et de la gigue, la qualité de service de la communication IAX avec les codecs G.711 et GSM est adéquate. Le résultat obtenu de la partie applicative nous a permis de prouver que le client mobile de vidéoconférence est capable de s’enregistrer auprès d’un Proxy/Registrar pour joindre une session multimédia et de signaliser avec d’autres clients de la session via le protocole SIP. La conception du protocole RTP dans la technologie mobile adopte le RFC 3250 sur le plan théorique. L’architecture du système utilisé et les composantes logicielles ont été bien mises en place. La transmission des paquets RTP a été bien réalisée. La manipulation des paquets RTP en mode binaire a été bien effectuée pour rediriger les flux audio et vidéo au lecteur JMStudio

Provisionamento em infraestruturas de voz sobre IP

Dissertação apresentado à Escola Superior de Tecnologia e Gestão do IPL para obtenção do grau de Mestre em Engenharia Informática - Computação Móvel, orientada pelo Professor Doutor Carlos Rabadão.Actualmente vive-se em plena era digital. Redes sociais, chamadas telefónicas pela Internet, telefones com vídeo-chamada, telemóveis e televisões com acesso à Internet, são tecnologias que não se podem ignorar. Assumem-se hoje como novas formas de comunicar, que aliam facilidade de utilização, maior alcance e custos mais reduzidos. Esta dissertação foca-se na tecnologia de telefonia utilizando uma rede de comutação de pacotes, como por exemplo, a Internet. Esta tecnologia denominada de voz sobre protocolo de Internet (VoIP), está hoje em grande expansão nas empresas e instituições devido, em parte, à redução de custos e aos novos serviços suportados. O aparecimento da telefonia IP trouxe inúmeras vantagens para o utilizador. Contudo, existe uma problemática associada, que são os custos de suporte e manutenção que esta tecnologia carece. Em tecnologias anteriores, os equipamentos telefónicos não necessitavam de configurações significativas, localmente em cada dispositivo, sendo praticamente tudo parametrizado na central telefónica. O mesmo não acontece em sistemas de VoIP. O Instituto Politécnico de Leiria (IPL) conta já com cerca de 300 telefones IP em produção e sempre que é preciso fazer algum tipo de manutenção em massa esta constitui-se uma actividade demorada, envolvendo muita mão-de-obra. Assim, neste projecto é proposta uma arquitectura que permita gerir todos estes equipamentos, em simultâneo, tendo em conta as funcionalidades de provisionamento suportadas por cada um. A arquitectura proposta baseia-se essencialmente em cinco fases – (i) provisionamento de um novo terminal, (ii) verificação de terminais registados no Asterisk, (iii) visualização de terminais adicionados à plataforma, (iv) alterações de configurações em massa, (v) actualização de firmware e (vi) tolerância a falhas. Para proceder à validação desta arquitectura foi desenvolvido um protótipo, tendo-se posteriormente efectuado a análise e avaliação do modelo proposto. Com esta respectiva análise e avaliação foi possível validar a arquitectura relativamente à sua utilidade para reduzir tempos em suporte e manutenção do sistema de telefonia IP do IPL