Title page, table of contents and abstract only. The complete thesis in print form is available from the University of Adelaide Library. / Current best-effort packet-switched Internet is not a perfect environment for real-time applications such as transmitting voice-over the network (Voice Over Internet Protocol or VOIP). Due to the unlimited concurrent access to the Internet by users, the packet loss problem cannot be avoided. Therefore, the VOIP based applications encompass problems such as "voice quality degradation caused by lost packets". The effects of lost packets are fundamental issues in real-time voice transmission over the current unreliable Internet. The dropped packets have a negative impact on voice quality and concealing their effects at the receiver does not deal with all of the drop consequences. It has been observed that in a very lossy network, the receiver cannot cope with all the effects of lost packets and thereby the voice will have poor quality. At this point the Active Networks, a relatively new concept in networking, which allows users to execute a program on the packets in active nodes, can help VOIP regenerate the lost packets, and improve the quality of the received voice. Therefore, VOIP needs special voice-packing methods. Based on the measured packet loss rates, many new methods are introduced that can pack voice packets in such a way that the lost packets can be regenerated both within the network and at the receiver. The proposed voice-packing methods could help regenerate lost packets in the active nodes within the network to improve the perceptual quality of the received sound. The packing methods include schemes for packing samples from low and medium compressed sample-based codecs (PCM, ADPCM) and also include schemes for packing samples from high compressed frame-based codecs (G.729). Using these packing schemes, the received voice has good quality even under very high loss rates. Simulating a very lossy network using NS-2 and testing the regenerated voice quality by an audience showed that significant voice quality improvement is achievable by employing these packing schemes. / http://proxy.library.adelaide.edu.au/login?url= http://library.adelaide.edu.au/cgi-bin/Pwebrecon.cgi?BBID=1147315 / Thesis (Ph.D.) -- University of Adelaide, School of Electrical and Electronic Engineering, 2004

The Datagram Congestion Control Protocol (DCCP) is a prominent transport protocol that has attracted the attention of the scientific community for its rapid progress and good results. The main novelty of DCCP is the performance priority design, as in UDP, however with congestion control capabilities, as in TCP. Literature about DCCP is still scarce and needs to be complemented to gather enouth scientific elements to support new research properly. In this context, this work joins the efforts of the scientific community to analise, mensure, compare and characterize DCCP in relevant scenarios that cover many real world situations. Three open questions were preliminarly identified in the literature: How DCCP behaves (i) when fighting for the same link bandwidth with other transport protocols; (ii) with highly relevant ones (e.g., Compound TCP, CUBIC) and (iii) fighting for the same link bandwidth with Compound TCP and CUBIC, adopting multimedia applications (e.g., VoIP). In this work, computational simulations are used to compare the performance of two DCCP variants (DCCP CCID2 and DCCP CCID3) with three highly representative TCP variants (Compound TCP, CUBIC and TCP SACK), in real world scenarios, including concurrent use of the same link by protocols, link errors and assorted bandwidths, latencies and traffic patterns. The simulation results show that, under contention, in most scenarios DCCP CCID2 has achieved higher throughput than Compound TCP or TCP SACK. Throughout the simulations there was a tendency of DCCP CCID3 to have lower throughput than the other chosen protocol. However, the results also showed that DCCP CCID3 has achieved significanly better throughput in the presence of link errors and higher values of latency and bandwidth, eventualy outperforming Compound TCP and TCP SACK. Finally, there was a tendency of predominance of CUBIC´ throughtput, which can be explained by its aggressive algorithm (i.e., non-linear) of return of the transmission window to the previous value before the discard event. However, CUBIC has presented the highest packet drop and the lowest delivery rate. / O Datagram Congestion Control Protocol (DCCP) é um proeminente protocolo de transporte que vem atraindo a atenção da comunidade científica pelos seus rápidos avanços e bons resultados. A principal inovação do DCCP é a priorização de desempenho, como ocorre com o UDP, mas com capacidade de realizar controle de congestionamento, como ocorre com o TCP. Entretanto, a literatura sobre o DCCP ainda é escassa e necessita ser complementada para trazer elementos científicos suficientes para novas pesquisas. Neste contexto, este trabalho vem se somar aos esforços da comunidade científica para analisar, mensurar, comparar e caracterizar o DCCP em cenários representativos que incorporem diversas situações de uso. Identificaram-se então três questões alvo, ainda em aberto na literatura: qual é o comportamento do DCCP (i) quando disputa o mesmo enlace com outros protocolos de transporte; (ii) com protocolos de transporte relevantes (e.g., Compound TCP, CUBIC) e (iii) em disputa no mesmo enlace com o Compound TCP e o CUBIC, utilizando aplicações multimídia (e.g., VoIP). Neste trabalho, simulações computacionais são utilizadas para comparar duas variantes do DCCP (CCID2 e CCID3) a três variantes do TCP (Compound TCP, CUBIC e TCP SACK), em cenários onde ocorrem situações de mundo real, incluindo utilização concorrente do enlace pelos protocolos, presença de erros de transmissão no enlace, variação de largura de banda, variação de latência, e variação de padrão e distribuição de tráfego. Os resultados das simulações apontam que, sob contenção, na maioria dos cenários o DCCP CCID2 obteve vazão superior à do Compound TCP, do DCCP CCID3 e do TCP SACK. Ao longo das simulações observou-se uma tendência do DCCP CCID3 a ter vazão inferior à dos demais protocolos escolhidos. Entretanto, os resultados apontaram que o DCCP CCID3 obteve desempenho significativamente melhor na presença de erros de transmissão e com valores maiores de latência e de largura de banda, chegando a ultrapassar a vazão do DCCP CCID2 e do TCP SACK. Por fim, observou-se uma tendência de predominância do protocolo CUBIC no tocante à vazão, que pode ser determinada pelo seu algoritmo agressivo (i.e., não-linear) de retorno da janela de transmissão ao valor anterior aos eventos de descarte. Entretanto, o CUBIC apresentou o maior descarte de pacotes e a menor taxa de entrega.

Multiprotocol Label Switching (MPLS) is an emerging technology which ensures the reliable delivery of the Internet services with high transmission speed and lower delays. The key feature of MPLS is its Traffic Engineering (TE), which is used for effectively managing the networks for efficient utilization of network resources. Due to lower network delay, efficient forwarding mechanism, scalability and predictable performance of the services provided by MPLS technology makes it more suitable for implementing real-time applications such as voice and video. In this thesis performance of Voice over Internet Protocol (VoIP) application is compared between MPLS network and conventional Internet Protocol (IP) network. OPNET modeler 14.5 is used to simulate the both networks and the comparison is made based on some performance metrics such as voice jitter, voice packet end-to-end delay, voice delay variation, voice packet sent and received. The simulation results are analyzed and it shows that MPLS based solution provides better performance in implementing the VoIP application. In this thesis, by using voice packet end-to-end delay performance metric an approach is made to estimate the minimum number of VoIP calls that can be maintained, in MPLS and conventional IP networks with acceptable quality. This approach can help the network operators or designers to determine the number of VoIP calls that can be maintained for a given network by imitating the real network on the OPNET simulator. / 0046737675303

Šio darbo tema - Internetinės telefonijos teisinis reglamentavimas Lietuvoje. Šiuolaikinis Internetinės telefonijos populiarumas ne tik sukelia vartotojų, bet ir reguliuotojų suinteresuotumą. Būtent ji pastaruoju metu kelia daugybę diskusijų tarptautiniuose bei nacionalinėse forumuose, kurių vienas pagrindinių aspektų – tinkamo Internetinės telefonijos reguliavimo sukūrimas. Pagrindinis baigiamojo magistrinio darbo tikslas – išnagrinėti Internetinės telefonijos reguliavimą tarptautiniu ir Lietuvos mastu, bei apžvelgti su juo susijusias problemas. Darbe nagrinėjama užsienio šalių praktika, remiantis kuria iškeliamos pagrindinės Lietuvos IP telefonijos teisinio reguliavimo gairės. Būtent: telefono numerių skyrimas, numerio perkeliamumas, skambučiai į pagalbos tarnybas, skambučiai kitais telefono numeriais, IP telefonijos skambučių saugumas, bei aprašomos su jų įgyvendinimu susijusios problemos. / The topic of the paper is Legal Reglamentation of VoIP Telephony in Lithuania. VoIP telephony’s nowadays spread and popularity scores an interest and debates not only between it’s consumers but also between legal regulators. VoIP is the main and rather often discussed topic of international and national forums, which aim to develop its proper regulation. The main aim of this paper is to internationally analyse VoIP’s legal regulation and to survey its associated problems. Foreign countries’ experience helps to formulate basic guidelines of Lithuanian VoIP legal regulation. Namely: numbering, numbers portability, calls to emergency services, calls to other telephone numbers, safety of VoIP calls and its associated problems.

The master’s thesis focuses on the virtual local computer network for laboratory usage. It aims to propose and realize proper network connection in order to monitor expected data flow. Thanks to the network analysers (software ClearSight and hardware NetTool Series II) it plans to pursue in detail the used transmission protocols of TCP/IP layers. The most decisive feature happens to be the right choice of appropriate network components and their precise configuration. Consequently, the thesis formulates a proposal of a laboratory task for the needs of students, which is also closely related to the actual problems. The assignment of the task will serve the teachers as a test pattern for measurement. The results elaborated in the form of the model protocol should enable later comparison of the recorded data. Another part of the diploma thesis is the working-out of well arranged manuals for the network analysers involved.