scholarly journals EVALUATION OF VOICE TRANSMISSION QUALITY IN THE LTE NETWORKS

Author(s):  
Ivan Vetoshko ◽  
Vyacheslav Noskov

Background. LTE mobile networks combine packet network technology and radio technology. Parameters of packet and radio subsystems significantly affects the quality of all traffic types transmission, especially telephone traffic, as the most demanding to such parameters of network transmission as delay, jitter and packet loss rate. The recommendations of the International Telecommunication Union and the documents of the partner organization of telecommunications operators (3GPP) contain hypothetical reference models, targets for end-to-end connection quality, and lists the factors that affect the quality (QoS) of VoLTE services. In addition, the network points are shown where you need to measure the quality of telephone traffic and tools for quality assessment. The quality of telephony services is assessed according to the E-model using the method of determining the mean opinion score (MOS). However, this technique is intended primarily to determine the MOS during the network planning. To calculate the MOS in a working network, you have to measure such network performance first such as voice delay and packet loss rate. This article presents the method of calculating MOS in the LTE network based on the E-model and presents the results of practical quality studies. Objective. The purpose of this article is research the impact of delay and packet loss ratio and voice codec characteristics in the real LTE network on quality of telephone services. Methods. Analysis of factors affecting on telephone services quality and analysis MOS assessment methods. Practical studies of the delay and packet loss ratio affect the MOS level in various conditions of radio coverage and network load. Results. Practical results of delay and packet loss ratio influence on the telephone services quality in the LTE network. Calculated MOS based on the practically measured delay and packet loss ratio. Conclusions. The combination of packet technologies, modern AMR-WB codecs and QoS support mechanisms in the LTE networks provides high quality perception of voice messages at the level of not less than 4 on the MOS scale. With a delay not exceeding 180 ms, a sufficiently high quality of voice transmission is ensured (MOS ≈ 4). VoLTE technology using the AMR-WB codec is quite resistant to packet loss and provides high quality perception of voice messages at a packet loss ratio of up to 1%.

TRANSIENT ◽  
2019 ◽  
Vol 7 (4) ◽  
pp. 971
Author(s):  
Danur Ilham Khoiruman ◽  
Sukiswo Sukiswo ◽  
Ajub Ajulian Zahra

Metro Ethernet merupakan salah satu teknologi untuk memberikan solusi terintegrasi untuk layanan suara, data dan video dalam cakupan yang luas (perkotaan). Teknologi ini memiliki kecepatan transmisi data sebesar 10 Mbps - 100 Gbps. Suatu jaringan harus memiliki kualitas layanan dan kapasitas yang memadai baik dari segi kapasitas link, router dan performansi QoS (Quality of Service). Pada penelitian ini, dirancang jaringan metro ethernet dengan kapasitas sesuai kebutuhan masyarakat kota Semarang tahun 2028 dengan kualias layanan yang sesuai dengan standar PT. Telkom dan ITU-T. Pemodelan dan pembuatan simulasi rancangan jaringan menggunakan perangkat lunak Riverbed Modeler 17.5. Perbandingan protocol routing RIP dan OSPF dengan parameter waktu konvergensi dilakukan sebelum analisis QoS, dengan tujuan mendapatkan rekomendasi protocol routing yang terbaik. Analisis parameter QoS yang diukur meliputi round trip delay (RTD), jitter, packet loss, utilisasi dan volume trafik. Hasil perbandingan protocol routing menunjukkan bahwa protocol routing OSPF memiliki waktu konvergensi lebih cepat 2 kali lipat dari protokol RIP. Hasil analisis QoS menyatakan bahwa QoS semua link telah sesuai dengan standar yang ada, nilai terbesar untuk RTD adalah 1,265 ms, untuk jitter adalah 0,7331 ms, untuk packet loss ratio adalah 0,00019214 %, untuk utilisasi tertinggi yaitu 58,4%, dan volume trafik terbesar adalah 91.636 Mbps.


2018 ◽  
Vol 10 (2) ◽  
pp. 191-201 ◽  
Author(s):  
Bambang Sugiantoro ◽  
Yuha Bani Mahardhika

ABSTRAK Performa layanan jaringan Internet pada UIN Sunan Kalijaga Fakultas Sains dan Teknologi masih belum maksimal, yaitu memiliki tingkat kualitas delay sebesar 159 milidetik menurut TIPHON Bagus. Besar Throughput sebesar 9.0 MBps dan presentase Throughput sebesar 50 % dikategorikan menurut standarisasi TIPHON sedang. Dan memiliki nilai packet loss ratio sebesar 36 % dikategorikan menurut standarisasi TIPHON adalah jelek. ABSTRACT Internet service network performance in Islamic State University of Sunan Kalijaga environment in the faculty of science and technology area is still not maximal. It has a delay quality level of 159 milliseconds according to good TIPHON. Large throughput of 9.0 Mbps and throughput percentage of 50% are categorized according to standardized of normal TIPHON and it has a value of packet loss ratio of 36% categorized according to TIPHON standardization is bad.How to Cite : Sugiantoro, B. Mahardhika, Y.B . (2017). ANALISIS QUALITY OF SERVICE JARINGAN WIRELESS SUKANET WiFi DI FAKULTAS SAINS DAN TEKNOLOGI UIN SUNAN KALIJAGA. Jurnal Teknik Informatika, 10(2), 191-201. doi:10.15408/jti.v10i2.7027Permalink/DOI: http://dx.doi.org/10.15408/jti.v10i2.7027


2012 ◽  
Vol 263-266 ◽  
pp. 1858-1863
Author(s):  
Jin He Zhou ◽  
Guo Min Xia

Diffserv-aware and Traffic Engineering combine the advantages of MPLS, Traffic Engineering (TE) and Differentiated Services (Diffserv, DS) to provide high performance and Quality of Service(QoS) in networks. We have designed three scenarios on Juniper Networks platforms to analyze the packet loss rate and delay for video, voice and data. The results show that MPLS DS-TE can improve the QoS for differentiated service effectively. The research has practical value for the development of DS-TE based on MPLS.


2013 ◽  
Vol 4 (1) ◽  
Author(s):  
Said Atamimi

Video streaming adalah aplikasi yang dapat melayani kebutuhan user akan data yang bersifat real time. Dengan adanya teknologi wireless LAN, user akan semakin dimudahkan dalam mengakses informasi seperti video streaming kapan saja dan di lokasi mana saja.Penelitian ini ditujukan agar dapat memperlihatkan hasil video streaming dari beberapa lokasi dalam lingkungan kantor Indosat. Dalam percobaan ini menggunakan beberapa perangkat antara lain satu buah server streaming, satu klien yang menggunakan laptop dan AP yang memang sudah ada dalam jaringan LAN Indosat serta skenario lokasi yang telah ditentukan sebagai tempat pengambilan data. Kemudian dilanjutkan pada tahap pengamatan sistem dengan melakukan peng-capture-an paket untuk mendapatkan data berupa throughput, delay, jitter, dan packet loss ratio dari tiap-tiap lokasi yang telah ditentukan.Hasil Penelitian ini, dengan adanya perbedaan lokasi mengakibatkan perbedaan dari kualitas video streaming berdasarkan parameter- parameter yang telah didapat pada percobaan.Kata Kunci : video streaming, wireless LAN, user, coverage


Author(s):  
Rui R. Paulo ◽  
Fernando J. Velez ◽  
Bahram Khan

This work aims at studying the indoor deployment of small cells, also known as femtocells, to provide coverage to a 5×5 grid geometry. The number of deployed HeNBs is 4, 5, or 6. An updated version of LTE-Sim is considered to extract values for Exponential Effective SINR Mapping (EESM), Packet Loss Ratio (PLR), maximum number of supported users, goodput and delay. Results reveal that the use of four HeNBs corresponds to the highest values of EESM. For the considered geometry, 3GPP suggested a maximum of five HeNBs. However, this deployment shows worser performance compared to the topology with four HeNBs. The geometry with six HeNBs is the one with the best overall performance results for the 5×5 grid of apartments.


2021 ◽  
Author(s):  
Colin Xialin Huang

There are increasing demands for real-time streaming video applications over the Internet. However, the current generation Internet was not originally designed for real-time streaming applications and only provides best-effort services, so there are many challenges in the deployment of video streaming applications over the Internet. This thesis investigates a hybrid end-to-end rate adaptation framework that provides application-level enhancements to achieve Quality of Service (QoS) for MPEG-4 FGS-Encoded video bandwidth on the path and the terminal process capabilities based on the packet-loss ratio and then determine their subscribing rate of video streams. The sender adjusts the transmission rate based on the packet-loss ratio and then determine their subscribing rate of video streams. The sender adjusts the transmission rate based on the proportion of load status feedbacks from the receivers. The sender and the receivers act together to minimize the possibility of network congestion by adjusting the transmission rate to match the network conditions. This framework achieves inter-receiver fairness in a heterogeneous multicast environment and improves QoS stability for MPEG-4 FGS video streaming over the Internet.


2013 ◽  
Vol E96.B (7) ◽  
pp. 1908-1917
Author(s):  
Takuya TOJO ◽  
Hiroyuki KITADA ◽  
Kimihide MATSUMOTO

Sign in / Sign up

Export Citation Format

Share Document