Study, Analysis and Modeling of IP Multimedia systems on next generation networks providing mobile and fixed multimedia services

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Sip-rtsp Convergence: Rtsp-c

Tez (Yüksek Lisans) -- İstanbul Teknik Üniversitesi, Fen Bilimleri Enstitüsü, 2008Thesis (M.Sc.) -- İstanbul Technical University, Institute of Science and Technology, 2008Bu çalışma ile SIP protokolü iletimde kullanılarak ve RTSP protokolü yetenekleri SIP mesaj gövdesine yerleştirilerek VOIP ağlarında yeni bir medya kontrol modeli öne sürülmüştür. RTSP-C olarak isimlendirilmiş olan bu yeni yakınlaşma modeli sadece SIP ve RTSP protokollerinin bir arada çalışmasını garantilemekle kalmamakta; aynı zamanda medya kontrol isteklerinin asıllanması ve oturum sunum bilgisi (SDP) alış verişindeki bazı açık noktalara çözüm getirmektedir. Bu yeni model RTSP protokolünün NAT geçirimie ait yöntemlere gereksinimini ortadan kaldırmakla beraber, SIP protokolünün NAT geçirim yöntemleri geçerliliğini korumaktadır. Bu modelin sağladığı asıl gelişme medya kontrolü bilgisi ve durum bilgisini SIP protokolüne açık hale getirmesidir. Bu sayede bu model medya yayınına dayalı yeni SIP servislerinin geliştirilmesine olanak sağlamaktadır. Bu proje kapsamında ortaya koyulan yeni modeli örneklendirmek amacıyla bir İsteğe Bağlı Görüntü Yayını (VoD) sistemi geliştirilmiştir. Bu uygulama ile RTSP-C yakınlaştırma modelinin çalışabilirliği doğrolanmıştır. Sonuçlar literatürdeki diğer örnekler ile karşılaştırıldığında modelin daha onceden belirlenen sorunlara uygun çözümleri sağladığı görülmüştür.In this study, using Session Initiation Protocol (SIP) as transport and placing Real Time Streaming Protocol (RTSP) capabilities in the SIP message body, a media control model has been introduced for Voice Over IP (VOIP) networks. This new convergence model does not only guarantee the interoperability of SIP and RTSP protocols but also resolve some open points on media control request authentication and session presentation (SDP) exchange. This new model is also valid for NAT Traversal methods applicable to SIP while it lifts the necessity of NAT Traversal methods for RTSP. The major advancement this model provides is: it makes the media control method/state information available to SIP. By doing that, this model enables the development of new streaming based SIP services. In this project content a Video on Demand (VoD) system is developped to instantiate the new convergence model. The implementation validated the operability of RTSP-C convergence model. The comparison of the results with other models on literature showed that the model provided adequate solutions on the pre-determined problems.Yüksek LisansM.Sc

Handling of IP-Addresses in the Context of Remote Access

Masteroppgave i informasjons- og kommunikasjonsteknologi 2008 – Universitetet i Agder, GrimstadFor various reasons (e.g., security, lack of IPv4-addresses) the services in the home smart space only use private IP addresses. This is unfortunate in the remote service access since these addresses frequently appear in responses sent from a service in the remote smart space (e.g., your home) to the visited smart space (e.g., your friend’s home).The Internet Engineering Task Force (IETF) provides some solutions and workarounds for the problem caused by NAT. In this project, the challenge to me is to summarize the available options, rank the options according to which one is preferred for the RA-scenario. I will come up with my practical NAT traversal techniques by testing and gathering data on the reliability of NAT traversal techniques since none of the existing ones seems to work well. A demonstration of the key features will be shown in the thesis. NAT traversal techniques apply to TCP and UDP need to be researched in advance. Handling of peers behind all kinds of NAT need to be tested and determined for the communication. The result of the paper will well improve the evaluation of specific issues on NAT and the creating of an UNSAF proposal

Network convergence and QoS for future multimedia services in the VISION project

The emerging use of real-time 3D-based multimedia applications imposes strict quality of service (QoS) requirements on both access and core networks. These requirements and their impact to provide end-to-end 3D videoconferencing services have been studied within the Spanish-funded VISION project, where different scenarios were implemented showing an agile stereoscopic video call that might be offered to the general public in the near future. In view of the requirements, we designed an integrated access and core converged network architecture which provides the requested QoS to end-to-end IP sessions. Novel functional blocks are proposed to control core optical networks, the functionality of the standard ones is redefined, and the signaling improved to better meet the requirements of future multimedia services. An experimental test-bed to assess the feasibility of the solution was also deployed. In such test-bed, set-up and release of end-to-end sessions meeting specific QoS requirements are shown and the impact of QoS degradation in terms of the user perceived quality degradation is quantified. In addition, scalability results show that the proposed signaling architecture is able to cope with large number of requests introducing almost negligible delay

Security for the signaling plane of the SIP protocol

VOIP protocols are gaining greater acceptance amongst both users and service providers. This thesis will aim to examine aspects related to the security of signaling plane of the SIP protocol, one of the most widely used VOIP protocols. Firstly, I will analyze the critical issues related to SIP, then move on to discuss both current and possible future solutions, and finally an assessment of the impact on the performance of HTTP digest authentication, IPsec and TLS, the three main methods use

Service provisioning in two open-source SIP implementation, cinema and vocal

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_363Adobe Acrobat 9.54 Paper Capture Plug-i