İnternet protokolü üzerinden ses iletimi

Abstract

Paket-temelli şebekelere dayalı telefon haberleşmesi yada Internet Protokolü (İP) üzerinden ses'li iletişim (Voice over IP - VoIP), aktarım ortamı interneti kullanarak ses iletişimini sağlar. İP üzerinden ses iletimi, geleneksel ses hizmetlerini veri şebekeleri üzerinden ulaştıran bütünleşik iletişim ortamını hedefliyor. Temelinde ses verileri kodlanarak, İP paketlerine yerleştirilir ve bir paket-temelli şebeke üzerinden iletilir. Geleneksel Kamu Bağlaşmalı Telefon Şebekesi (Public Switched Telephone Network - PSTN) ile internet gibi paket şebekelerini birleştiren ve özel bir işlevselliğe sahip olan bu çözüm, servis sağlayıcılara yönelik internet üzerinden sesli iletişim sunucusu donanım ve yazılımına dayanır. Bu teknoloji ile kullanıcılar, PSTN ve IP-temelli veri şebekeleri (ör: ATM, FR v.b.) üzerinden uygun, kesintisiz, düşük maliyetli ve güvenilir koşullarda ses mesajı trafiği olanağına kavuşacaklar. Bu platform, internet'te telefondan telefona konuşmayı, fakstan faksa iletişimi ve internet telefonu arabağdaşımlarını içermektedir. Bugün haberleşme standartlarını geliştiren iki teşkilatın "İP üzerinden ses iletişimi" konusunda yoğun çalışmaları yürütülmektedir. Aynı işlevselliğe sahip bu iki çözümün ilki ITU-T (International Telecommunication Union) tarafından önerilen H.323 tavsiyesi, ikincisi ise lETF'nin (Internet Engineering Task Force) yayınladığı MGCP (Media Gateway Control Protocol) protokol bütünlüğüdür. Bu bitirme çalışmasında farklı şebekeler üzerinde çoğul-ortam uygulamalarına esas teşkil eden ve ITU-T'nin diğer H.32x1 tavsiyelerinin bir parçası olan H.323 standardı ele alınıp açıklanmaya çalışılmıştır. 1 H.323 standardı, ITU-T'nin tanımladığı H.32x aile tavsiyelerinin bir öğesidir. Geri kalan çoğul- ortam haberleşme öğeleri aşağıdakilerdir:. Devre Bağlaşmalı Şebekeler (Switched Circuit Network - SCN) üzerinde H.324 standardı,. Tümleşik Hizmetler Sayısal Şebeke (Integrated Services Digital Network - ISDN) üzerinde H.320 standardı,. Genişbandlı Tümleşik Hizmetler Sayısal Şebeke (Broadband ISDN) üzerinde H.321 ve H.310 standartları,. Kalite hizmetini (QOS) temin eden LAN üzerinde H.322 standardı.

Data traffic has traditionally been forced to fit onto the voice network. The Internet has created an opportunity to reverse this integration strategy - voice and facsimile can now be carried over IP networks, with the integration of video and other multimedia applications close behind. The Internet and its underlying TCP/IP protocol suite has become the driving force for new technologies, with the unique challenges of real-time voice being the latest in a series of developments. Support for voice communications using the Internet Protocol (IP), which is usually just called "Voice over IP" or VoIP, has become especially attractive given the low-cost, flat rate pricing of the public Internet. VoIP can be defined as the ability to make telephone calls (i.e., to do everything we can do with the today's PSTN) and to facsimiles over IP- based data networks with suitable quality of service (QoS). Many industry analysts estimate that the overall VoIP market will become a multi-billion dollar business within three years. The H.323 standard provides a foundation for audio, video and data communications across IP-based networks, including the Internet. H.323 is an "Umbrella Recommendation" from the ITU-T that sets standards for multimedia communications over Local Area Networks. H.323 is part of a family of ITU-T recommendations called H.32x that provides multimedia communications services over variety of networks. Hence, H.323 is a base for Voice over IP technology. Voice over packet transfer can significantly reduce the perminute cost, resulting in reduced long-distance bills. In fact, many dial-around-calling schemes available today already rely on VoIP backbones to transfer voice, passing some of the cost savings to the customer. These high-speed backbones take advantage of the convergence of Internet and voice traffic to form a single managed network. This network convergence also opens the door to novel applications. Interactive shopping (web pages incorporating a "click-to-talk" button) is just one example, while streaming audio, electronic white-boarding and CD-quality conference calls in stereo are other exciting applications. VIII. The main justifications for development of VoIP can be summarized as follows:. Cost Reduction There can be real savings in long distance telephone costs, which is extremely important to most companies, particularly those with international markets.. Simplification An integrated voice/data network allows more standardization and reduces total equipment needs.. Advanced Applications The long run benefits of VoIP include support for multimedia and multiservice applications, something that today's telephone system can't complete with. But along with the initial excitement, users are worried over possible degradation in voice quality when voice is carried over these packet networks. Whether these concerns are based on experience with the early Internet Telephony applications, or whether they are based on understanding the nature of packet networks, voice quality is a critical parameter in acceptance of VoIP services. As such, it is crucial to understand the factors affecting voice over packet transmission, as well as obtain the tools to measure and optimize them. The thesis states on the factors that will probably reduce the performance of VoIP. The thesis discusses the Voice over IP technology within H.323 protocol standard. H.323 is explained with an emphasis on Gateways and Gatekeepers, which are component of an H.323 network. The call flows between entities in an H.323 network are explained. The thesis defines the functions of two closely linked parts. Namely, Real-Time Transport Protocol (RTP) and Real-Time Transport Control Protocol (RTCP). RTP provides end-to-end network transport functions suitable for applications transmitting real-time data such as audio, video or data, over multicast or unicast network services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery. The RTCP is based on the periodic transmission of control packets to all participants in the session, using the same distribution mechanism as the data packets. RTP and RTCP are designed to be independent of the underlying transport and networks layers. In order to clarify the RTP specification, implementation codes in C language are provided for RTP and RTCP message formats. These implementation notes are for informational purposes only.