Since the late 90's IP telephony, commonly referred to as Voice over IP (VoIP), has been presented as a revolution on communications enabling the possibility to converge historically separated voice and data networks, reducing costs, and integrating voice, data and video on applications. This paper presents a study over the standard VoIP protocols H.323, session initiation protocol (SIP), media gateway control protocol (MGCP), and H.248/Megaco. Given the fact that H.323 and SIP are more widespread than the others, we focus our study on them. For each of these protocols we describe and discuss its main capabilities, architecture, stack protocol, and characteristics. We also briefly point their technical limitations. Furthermore, we present the advanced multimedia system (AMS) project, a new system that aims to operate on Next Generation Networks (NGN) taking the advantage of its features, and it is viewed as the successor to H.323 and SIP.
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IEEE Communications Surveys & Tutorials
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Currently two standards exist for signaling and control of voice over IP calls, namely ITU-T Recommendation H.323 and the IETF Session Initiation Protocol (SIP). Although there are a significant number of similarities between these two protocols, they behave significantly different when providing different higher level services. On one hand, SIP provides flexibility with broader scope, offering functions specifically designed to
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Voice will remain a fundamental communication media that cuts across people of all walks of life. It is therefore important to make it cheap and affordable. To be reliable and affordable over the common Public Switched Telephone Network, change is therefore inevitable to keep abreast with the global technological change. It is on this basis that this paper tends to critically review this new technology VoIP, x-raying the different types. It further more discusses in detail the VoIP system, VoIP protocols, and a comparison of different VoIP protocols. The compression algorithm used to save network bandwidth in VoIP, advantages of VoIP and problems associated with VoIP implementation were also critically examined. It equally discussed the trend in VoIP security and Quality of Service challenges. It concludes by reiterating the need for a cheap, reliable and affordable means of communication that would not only maximize cost but keep abreast with the global technological change.
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This is a review papers aims to discuss the various issues related to Session Initiation protocol, starting with the basic structure to the future aspects. Discussion includes Signaling, Me ssages, characteristics, extension, security and Quality of Services. Session Initiation Protocol provides advanced functional ities for signaling and control for Multimedia services. Application of Session Initiation Protocol ranges from Internet telephony and video conferencing to instant messaging and network services.
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Nowadays information technology, especially the Internet developed very rapidly, which is actually a Internet computers connected to each other. Telephony technology is also developed very fast and there is some alternative to use VoIP beside analog telephone because the cost is cheaper. VoIP also use codec that can compress voice data but the quality is still good. This research design an open source system of Asterisk server because company need of VoIP that can support traditional analog telephony system. Beside design an open source system, some codec technology is also tested, which are G.711 as commonly codec and also G.729 and G.723.1 as propiteary codecs, offering less bandwidth and more clearly sound than G.711. G.729 and G.723.1 is limited for one user only so it can be tested only for one user. After codec testing is arranged then an interconnection system of PSTN or analog telephony system is also tested. Using Linksys SPA-3102 interconnection to analog telephony is also tested and worked for one client. Abstrak Saat ini teknologi informasi, terutama Internet berkembang sangat pesat, sehingga ada teknologi jaringan internet yang saling menghubungkan komputer tersebut. Teknologi telephony juga berkembang sangat cepat dan ada beberapa alternatif untuk menggunakan VoIP disamping telepon analog karena biayanya lebih murah. VoIP menggunakan codec yang bisa mengkompresi data suara namun kualitasnya tetap bagus. Penelitian ini merancang sistem open source server Asterisk karena perusahaan membutuhkan VoIP yang dapat mendukung sistem telepon analog. Selain merancang sistem open source, beberapa teknologi codec juga diuji, yaitu G.711 sebagai codec yang berlaku umum dan juga G.729 dan G.723.1 sebagai codec propiteary, yang menawarkan bandwidth lebih sedikit dengan suara yang lebih jelas daripada G.711. G.729 dan G.723.1 terbatas hanya untuk satu pengguna sehingga hanya bisa diuji untuk satu pengguna saja. Setelah pengujian codec dilakukan maka sistem interkoneksi PSTN atau sistem telepon analog juga diuji. Interkoneksi dilakukan dengan voice gateway Linksys SPA-3102 dihubungkan ke telepon analog juga diuji dan dilakukan untuk satu klien.
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Voice over Internet Protocol (VoIP) has been widely deployed since the integration of the voice and data networks reduces management effort and cost. Since VoIP share the same infrastructure with traditional data network, it inherits all security problems from data network. Furthermore, VoIP also has its own security problems coming from new protocols and network component. This paper focuses on these VoIP specific security threats and the countermeasures to mitigate the problem. At first, this paper gives a brief introduction of VoIP techniques: the network structure, network components, protocols and standards, data handling procedures, quality of service requirements. Secondly, the paper discusses the VoIP specific security threats using the principle of CIA (Confidentiality, Integrity and Availability). The countermeasure to mitigate these threats is also discussed. At last, the paper proposes the practice to secure VoIP networks.
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