Security aspects in voice over IP systems

Security has become a major concern with the rapid growth of interest in the internet. This project deals with the security aspects of VoIP systems. Various supporting protocols and technologies are considered to provide solutions to the security problems. This project stresses on the underlying VoIP protocols like Session Initiation Protocol (SIP), Secure Real-time Transport Procotol (SRTP), H.323 and Media Gateway Control Protocol (MGCP). The project further discusses the Network Address Translation (NAT) devices and firewalls that perform NAT. A firewall provides a point of defense between two networks. This project considers issues regarding the firewalls and the problems faced in using firewalls for VoIP; it further discusses the solutions about how firewalls can be used in a more secured way and how they provide security

VoIP under the EU regulatory framework : preventing foreclosure?

In June 2004, the European Commission (EC) issued an "Information and Consultation Document" (European Commission 2004) that discussed how the Regulatory Framework of the European Union (EU) should be adapted to accommodate Voice over IP (VoIP) and invited relevant parties to comment on the Consultation Document. In our study, we use the responses of the different market parties to identify how incumbents seek to foreclose the market for VoIP telephony. From these responses we conclude that foreclosure is not only attempted by setting high prices for the use of infrastructure, but also by the strategic choice of infrastructure technology, which raises the cost of entry. We label the latter form of foreclosure "technological foreclosure" – as opposed to "market foreclosure". A simple modeling exercise shows that regulators seeking to avoid market foreclosure might trigger technological foreclosure. We argue that this has happened with the unbundling of the local loop in the EU, and that it might happen again with the transition to VoIP. We conclude that the current rights and obligations assigned to telecom companies effectively protect incumbents from competition by VoIP entrants. Moreover, the inaction of regulatory authorities when it comes to numbering and communication protocols is advantageous for incumbents and might obstruct the provision of new services in the future

Sip Based Mobile Voice Over Ip Client For Wireess Networks

Tez (Yüksek Lisans) -- İstanbul Teknik Üniversitesi, Fen Bilimleri Enstitüsü, 2008Thesis (M.Sc.) -- İstanbul Technical University, Institute of Science and Technology, 2008Bu tez SIP tabanlı mobile bir VoIP istemcisinin tasarımını ve gerçeklenmesini tanımlar. Bu tez temelde çoktürel ağlar üzerinde çalışabilen bir VoIP istemcisi tasarımının çözülmesi gereken iki sorununun üzerinde yogunlaşır. Birinci ve en zorlu sorun farklı erişim teknolojileri arasında kullanıcıya fark ettirmeden yer değişim desteği sağlanmasıdır. Bu tezde, kullanıcıya fark ettirmeden el değiştirme yönetimi, uygulama katmanında, multimedya oturumunu başlatmak, sonlandırmak ve değiştirmek için kullanılan Oturum Başlatma Protokolü (SIP) kullanılarak ele alınmıştır. SIP yaygın bir şekilde kabul edilmekte olan bir VoIP standartıdır. Kullanıcıya fark ettirmeden el değiştirmeyi destekleyebilmek için, VoIP istemcisi üzerinde çalışan SIP tabanlı bir bağlantı yöneticisi önerilmiştir. Bağlantı yöneticisi yeni ağlar keşfettiğinde, adaylar listesinden bir ağ seçer ve hali hazırda yürütülmekte olan iletişimi kullanıcıya fark ettirmeden yeni ağ arayüzüne aktarır. Dolayısı ile, bu birim Wi-Fi, 3G gibi çoktürel ağlar arasında dolaşmayı sağlar. İkinci sorun ise, en kaliteli çağrı (arama) desteğini sağlamaktır. En kaliteli çağrı desteği, iletişim kurmak isteyen tarafların farklı türden ağlara bağlı olmaları durumunda, VoIP uygulamasının iletişim tipine (yarı-çift yönlü yada tam-çift yönlü) karar vermebilmesi demektir. Örneğin, eğer iletişim kurmak isteyen taraflardan biri bir GSM ağındaysa, en iyi çağrı kalitesini yakalayabilmek için, iletişim yarı-çift yönlü olarak kurulmalıdır. Bu tez, bahsedilen özelliği desteklemek için, istemci tabanlı bir karar mekanizması önerir. Bu karar mekanizması, iletişim kurulmak istenen tarafa, istemcinin içinde bulunduğu ağa göre belirlenmiş iletişim tipini içeren bir davet iletisi gönderir. Diğer istemci bu davet iletisini aldıktan sonra, aynı karar mekanizması, iletişimi “bas-konuş VoIP” yada “tam-çift yönlü VoIP” olarak ayarlar.This thesis describes the design and the implementation of a SIP-based mobile VoIP client. It mainly focuses on two challenges of designing a VoIP client which works on heterogeneous network environments. One and the most challenging problem is the provision of seamless mobility support among different access technologies. In this thesis, seamless handover management is handled at the application layer by using Session initiation protocol (SIP), which is used to initiate, terminate, and modify multimedia session. SIP is becoming a widely accepted standard for VoIP. To support seamless handover, a SIP based connection manager is proposed on VoIP client application. As new networks are discovered by the connection manager, it selects a new network from the candidate list and transfers the current communication to the new network interface seamlessly. Therefore, this module provides roaming across heterogeneous networks such as Wi-Fi, 3G. Second problem is providing the best effort call quality support. It means that if the communication parties are in dissimilar networks, the VoIP application should decide the communication type (half-duplex or full-duplex). For instance, if one of the communication parties is in a GSM network, then the communication should be established as a half-duplex manner to achieve best call quality. This thesis proposes a client-based decision mechanism to support this property. This decision mechanism sends an invite message including the communication type (half-duplex or full-duplex) of the client according to the network in which it operates to the other communication party. After the other client receives this invite message, same decision mechanism adjusts the communication as either a “push to talk VoIP” or a “full-duplex VoIP”.Yüksek LisansM.Sc

Analyzing Voice And Video Call Service Performance Over A Local Area Network

Tez (Yüksek Lisans) -- İstanbul Teknik Üniversitesi, Fen Bilimleri Enstitüsü, 2010Thesis (M.Sc.) -- İstanbul Technical University, Institute of Science and Technology, 2010Bu çalışmada, VOIP teknolojisinden ve bu teknolojiyi kablolu ve kablosuz ortamda gerçeklemenin en önemli darboğazları anlatılacaktır. Ayrıca H.323, SIP (Session Initiation Protocol), Megaco ve MGCP gibi yaygın olarak kullanılan ses iletim protokolleri ve H.261, H.263 ve H.264 gibi görüntü iletim protokollerinden bahsedilmiştir. Ses kodek seçimi ve VOIP servis kalitesine etki eden faktörleri anlatılmaktadır. Bu tezde, ses, görüntü ve veri iletişimini aynı anda bünyesinde barındıran gerçek şebekeler simüle edilecektir. Kullanıcılara rastlantısal olarak ses, görüntü ve FTP gibi birtakım uygulamalar atanmıştır. Ayrıca önerilen kablolu şebekeye, kablosuz bir şebeke ilave edilerek sonuçlar incelenecektir. Optimal servis kalitesini sağlamak için seçilen uygun kuyruklama mekanizmaları ve kodek seçimlerini içeren senaryolar incelenecek ve OPNET ile elde edilmiş simülasyon sonuçları tartışılacaktır.In this study, we present a detailed description of the VoIP and also the most common challenges of implementing voice communication into wireline or wireless networks are discussed. Common voice protocols, such as H.323, Session Initiation Protocol (SIP), Megaco, MGCP and video protocols such as H.261, H.263, H.264 are described as well. CODEC selection and factors affecting VoIP Quality of Service are analyzed. We simulate a real network which includes both voice, video and data communication simultaneously. Workstations are randomly assigned to different applications, such as voice, video and FTP. We will also implement a wireless network to our proposed system. The scenarios including selecting appropriate queuing scheme and codec selection are presented and the simulation results with OPNET are drawn.Yüksek LisansM.Sc

Improving the robustness of CELP-like speech decoders using late-arrival packets information : application to G.729 standard in VoIP

L'utilisation de la voix sur Internet est une nouvelle tendance dans Ie secteur des télécommunications et de la réseautique. La paquetisation des données et de la voix est réalisée en utilisant Ie protocole Internet (IP). Plusieurs codecs existent pour convertir la voix codée en paquets. La voix codée est paquetisée et transmise sur Internet. À la réception, certains paquets sont soit perdus, endommages ou arrivent en retard. Ceci est cause par des contraintes telles que Ie délai («jitter»), la congestion et les erreurs de réseau. Ces contraintes dégradent la qualité de la voix. Puisque la transmission de la voix est en temps réel, Ie récepteur ne peut pas demander la retransmission de paquets perdus ou endommages car ceci va causer plus de délai. Au lieu de cela, des méthodes de récupération des paquets perdus (« concealment ») s'appliquent soit à l'émetteur soit au récepteur pour remplacer les paquets perdus ou endommages. Ce projet vise à implémenter une méthode innovatrice pour améliorer Ie temps de convergence suite a la perte de paquets au récepteur d'une application de Voix sur IP. La méthode a déjà été intégrée dans un codeur large-bande (AMR-WB) et a significativement amélioré la qualité de la voix en présence de <<jitter » dans Ie temps d'arrivée des trames au décodeur. Dans ce projet, la même méthode sera intégrée dans un codeur a bande étroite (ITU-T G.729) qui est largement utilise dans les applications de voix sur IP. Le codeur ITU-T G.729 défini des standards pour coder et décoder la voix a 8 kb/s en utilisant 1'algorithme CS-CELP (Conjugate Stmcture Algebraic Code-Excited Linear Prediction).Abstract: Voice over Internet applications is the new trend in telecommunications and networking industry today. Packetizing data/voice is done using the Internet protocol (IP). Various codecs exist to convert the raw voice data into packets. The coded and packetized speech is transmitted over the Internet. At the receiving end some packets are either lost, damaged or arrive late. This is due to constraints such as network delay (fitter), network congestion and network errors. These constraints degrade the quality of speech. Since voice transmission is in real-time, the receiver can not request the retransmission of lost or damaged packets as this will cause more delay. Instead, concealment methods are applied either at the transmitter side (coder-based) or at the receiver side (decoder-based) to replace these lost or late-arrival packets. This work attempts to implement a novel method for improving the recovery time of concealed speech The method has already been integrated in a wideband speech coder (AMR-WB) and significantly improved the quality of speech in the presence of jitter in the arrival time of speech frames at the decoder. In this work, the same method will be integrated in a narrowband speech coder (ITU-T G.729) that is widely used in VoIP applications. The ITUT G.729 coder defines the standards for coding and decoding speech at 8 kb/s using Conjugate Structure Algebraic Code-Excited Linear Prediction (CS-CELP) Algorithm

Voice over IP

The area that this thesis covers is Voice over IP (or IP Telephony as it is sometimes called) over Private networks and not over the Internet. There is a distinction to be made between the two even though the term is loosely applied to both. IP Telephony over Private Networks involve calls made over private WANs using IP telephony protocols while IP Telephony over the Internet involve calls made over the public Internet using IP telephony protocols. Since the network is private, service is reliable because the network owner can control how resources are allocated to various applications, such as telephony services. The public Internet on the other hand is a public, largely unmanaged network that offers no reliable service guarantee. Calls placed over the Internet can be low in quality, but given the low price, some find this solution attractive. What started off as an Internet Revolution with free phone calls being offered to the general public using their multimedia computers has turned into a telecommunication revolution where enterprises are beginning to converge their data and voice networks into one network. In retrospect, an enterprise\u27s data networks are being leveraged for telephony. The communication industry has come full circle. Earlier in the decade data was being transmitted over the public voice networks and now voice is just another application which is/will be run over the enterprises existing data networks. We shall see in this thesis the problems that are encountered while sending Voice over Data networks using the underlying IP Protocol and the corrective steps taken by the Industry to resolve these multitudes of issues. Paul M. Zam who is collaborating in this Joint Thesis/project on VoIP will substantiate this theoretical research with his practical findings. On reading this paper the reader will gain an insight in the issues revolving the implementation of VoIP in an enterprises private network as well the technical data, which sheds more light on the same. Thus the premise of this joint thesis/project is to analyze the current status of the technology and present a business case scenario where an organization will be able to use this information

Decorating Asterisk : experiments in service creation for a multi-protocol telephony environment using open source tools

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