Analisis Perbandingan Unjuk Kerja Protokol Udp Dengan DCCP Menggunakan Trafik Data Multimedia (original) (raw)
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Analysis of Datagram Congestion Control Protocol (DCCP
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Performance evaluation of the DCCP protocol in high-speed networks
The Internet transport layer protocols, TCP and UDP, do not provide an efficient transport for multimedia streams. In despite of that, UDP is often employded for such type of application, due to its low overhead. Since UDP does not implement any kind of congestion control mechanism, it has the potential of hindering Internet operability. To address such issue, a novel transport layer protocol, namely DCCP, has been proposed. It is designed to be used by multimedia applications with the aim of replacing the UDP protocol. This paper conducts a performance evaluation of the DCCP protocol in high-speed networks. The aim is to evaluate the scalability, the fairness, the convergence and the compatibility with TCP-Reno.
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WSEAS TRANSACTIONS on …, 2008
Delivering real-time video over the Internet is an important issue for many Internet multimedia applications. Transmission of real-time video has bandwidth, delay, and loss requirements. The applicationlevel quality for video streaming relies on continuous playback, which means that neither buffer underflow nor buffer overflow should occur. Since the Best Effort network such as the Internet does not provide any Quality of Service (QoS) guarantees to video transmission over the Internet. Thus, mapping the application-level QoS requirements into network-level requirements, namely, limited delay jitters. End-to-end application level QoS has to be achieved through adaptation. Since the QoS of video streams over IP networks depends on several factors such as video transmission rate, packet loss rate, and end-to-end transmission delay. The objectives of this paper are to simulate an adaptation scheme to include the effect of User Datagram Protocol (UDP) parameters on delay jitter and datagram loss values to increase the efficiency of UDP protocol to prevent the network congestion and increase the adaptivity and also, simulate an adaptation scheme to include the effect of Transmission Control Protocol (TCP) parameters on the transmission rates to increase the adaptivity of the transmission.
Transient, 2013
Session Initiation Protocol (SIP) merupakan protokol standar multimedia sebagai produk dari Internet Engineering Task Force (IETF) dan telah digunakan sebagai standar VoIP. Protokol ini menggabungkan teknologi seluler dan dunia internet. Sebuah sesi dalam jaringan SIP dapat berupa panggilan suara, e-mail, pesan teks, atau video streaming. Keunggulan SIP adalah operator jaringan dapat menggunakannya untuk mengontrol semua bentuk komunikasi dalam jaringan, bukan hanya suara. Protokol SIP berasal dari Simple Mail Transport Protocol (SMTP) dan Hypertext Transport Protocol (HTTP). Pada penelitian ini layanan Video Streaming, Audio Streaming, Voice over IP (VoIP) dan Conference Call dalam jaringan lokal SIP menggunakan Graphical Network Simulator (GNS3) dianalisis dengan mengamati Quality of Service (QoS) yaitu nilai delay, jitter, packet loss dan throughput dengan Wireshark. Dilakukan juga variasi faktorfaktor yang mempengaruhi QoS yaitu variasi jenis file, ukuran file, panjang kabel Lokal Area Network (LAN) dan gangguan pada pemutaran/playback video dan audio streaming. Dari hasil pengujian yang dilakukan, diperoleh bahwa pada layanan video dan audio streaming, QoS/kinerja layanan dipengaruhi oleh jenis file, ukuran file dan gangguan pada playback. QoS video dan audio streaming secara umum sudah memenuhi standar ITU G.114 baik dilihat dari besar delay, jitter, packet loss maupun throughput-nya. Pada layanan VoIP dan Conference Call, QoS dipengaruhi oleh variasi panjang kabel LAN dan secara umum sudah memenuhi standar.
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In recent years, usage of Mobile Ad-hoc Networks (MANETs) for communication has grown at a faster rate due to its ease of implementation and flexibility. Also,transmission of multimedia contents over Internet isone of the most widely used technologies being used globally. According to the ongoing trends in technology, most of the contents (data) sent over the Internet are interactive multimedia contents, which prefer to be delivered in error-state than being discarded or arriving late.To avoid network congestion, it is preferred to transmit the data without any overhead of prior connection establishment. A solution to both the problems is to use UDP as transport protocol, which provide no reliability and have low protocol processing overhead. An enhanced version of UDP, called UDP-Lite was also introduced a decade ago, which has been specifically designed for real-time multimedia applications. The aim of this paper is to compare the performances of UDP and UDP-Lite by changing various network parametersfor transmitting various video codecs.
Performance Evaluation of TCP, UDP and DCCP Traffic Over 4G Network
Fourth Generation (4G) mobile systems has been used more widely than the older generations 3G and 2G. Among the reasons are that the 4G’s transfer rate is higher and it supports all multimedia functions. Besides, its’ supports for wide geographical locus makes wireless technology gets more advanced. The essential goal of 4G is to enable voice-based communication being implemented endlessly. This study tries to evaluate if the old protocols suit with this new technology. And which one has the best performance and which one has the greatest effect on throughput, delay and packet loss. The aforementioned questions are crucial in the performance evaluation of the most famous protocols (particularly User Datagram Protocol (UDP), Transmission Control Protocol (TCP) and Datagram Congestion Control Protocol (DCCP)) within the 4G environment. Through the Network Simulation-3 (NS3), the performance of transporting video stream including throughput, delay, packet loss and packet delivery ratio are analyzed at the base station through UDP, TCP and DCCP protocols over 4G’s Long Term Evaluation (LTE) technology. The results show that DCCP has better throughput and lesser delay, but at the same time it has more packet loss than UDP and TCP. Based on the results, DCCP is recommended as a transport protocol for real time video.
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User Datagram Protocol (UDP) and Transmission Control Protocol (TCP) are a transportation layer routing protocols which are considered of the core protocols of the internet protocol suite. The behaviour of these routing protocols with different network metrics and scenarios is still not very clear. Therefore, this paper presents a comparison of the performance of both TCP and UDP to precisely determine which of these protocols is better. Network Simulator version 2.35 (NS2) is utilized to analyse and evaluate the performance for both TCP and UDP protocols varying in the packet size and the bandwidth. In this study, we have used two scenarios, in the first scenario the bandwidth has been changed with fixed packet size and in the second scenario the packet size has been changed with fixed bandwidth to precisely verify the performance of these protocols. These protocols were examined in terms of the rate end-to-end delay, rate throughput, packet delivery ratio, and packet loss ratio
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IEEE INFOCOM 2007 - 26th IEEE International Conference on Computer Communications, 2007
Most Internet telephony applications currently use either TCP or UDP to carry their voice-over-IP (VoIP) traffic. This choice can be problematic, because TCP is not well suited for interactive traffic and UDP is unresponsive to congestion. The IETF has recently standardized the new Datagram Congestion Control Protocol (DCCP). DCCP has been designed to carry media traffic and is congestion-controlled. This paper experimentally evaluates the voice quality that Internet telephony calls achieve over prototype implementations of basic DCCP and several DCCP variants, under different network conditions and with different codecs. It finds that the currently-specified DCCP variants perform less well than expected when compared to UDP and TCP. Based on an analysis of these results, the paper suggests several improvements to DCCP and experimentally validates that a prototype implementation of these modifications can significantly increase voice quality.
The effects of inter-packet spacing on the delivery of multimedia content
Proceedings 21st International Conference on Distributed Computing Systems, 2001
Streaming multimedia content with UDP has become increasingly popular over distributed systems such as the Internet. However, because UDP does not possess any congestion-control mechanism and most best-effort traffic is served by the congestion-controlled TCP, UDP flows steal bandwidth from TCP to the point that TCP flows can starve for network resources. Furthermore, such applications may cause the Internet infrastructure to eventually suffer from congestion collapse because UDP traffic does not selfregulate itself. To address this problem, next-generation Internet routers will implement active queue-management schemes to punish malicious traffic, e.g., non-adaptive UDP flows, and to the improve the performance of congestioncontrolled traffic, e.g., TCP flows. The arrival of such routers will cripple the performance of today's UDP-based multimedia applications. So, in this paper, we introduce the notion of inter-packet spacing with control feedback to enable these UDP-based applications to perform well in the next-generation Inter
Prociding, 2019
Quality measurement needs to be done to determine the level of quality possessed by the system. Quality of Service is a measurement method issued by the European Telecommunications Standard Institute (ETSI). All activities at AMIK DCC utilize internet facilities, such as teaching and learning activities, administrative activities, student unit activities, and libraries. Because to support all these activities need a good and reliable of network. Some students complained about the lack of satisfaction of the internet network on campus.this research will analyze the Quality of Service internet network both wired or wireless in AMIK DCC Building by using Internet Control Message Protocol (ICMP). The data obtained will then be processed for analyzed throughput, packet loss, delay, and jitter to know how the quality of internet services. Based on the measurement, the results obtained that the throughputis 79%, the packet loss under 5%, the delay is below 175 ms and the jitter at 1%. From the results of research, it can be concluded that the Quality Of Service internet network in AMIK DCC included in the category satisfactory by TIPHON standards, but need to add repeaters to defeat the Internet network coverage for every floor and room. 1. Introduction Computer network technology has penetrated into various fields and aspects of life. This can be seen from the use of computer networks by agencies, groups and individuals. Computer network technology is very important, because the many advantages possessed include easy and efficient process. AMIK Dian Cipta Cendikia provides internet access and facilities for a number of access points that are spread out at certain points so that internet connections are spread throughout all campus area. However, bandwidth management implemented in Shared Unlimited results in "grabbing" bandwidth so that internet access becomes unstable and sometimes even slow when there are many users accessing the same access point at the same time and vice versa. In addition, sometimes WLAN networks cannot be used to access the internet. In some locations there are also areas that have not been covered by WLAN networks or blind spots. From these statements, to be able to maintain the quality of service performance on the WLAN network in Dian Cipta Dian Cendikia always in good performance. It is necessary to monitor and analyze the WLAN network quality of service to minimize and find out network disruptions early. So that the WLAN network can always in maximum performance to be able to support ICT-based education services.Quality of Service (QoS) is a method of measuring how well a network is and an attempt to