voice transmission
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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%.


2021 ◽  
Vol 10 (6) ◽  
pp. 3202-3210
Author(s):  
Sameer A. S. Lafta ◽  
Mohaned Mahdi Abdulkareem ◽  
Raed Khalid Ibrahim ◽  
Marwah M. Kareem ◽  
Adnan Hussein Ali

The universal mobile telecommunications system (UMTS) has distinct benefits in that it supports a wide range of quality of service (QoS) criteria that users require in order to fulfill their requirements. The transmission of video and audio in real-time applications places a high demand on the cellular network, therefore QoS is a major problem in these applications. The ability to provide QoS in the UMTS backbone network necessitates an active QoS mechanism in order to maintain the necessary level of convenience on UMTS networks. For UMTS networks, investigation models for end-to-end QoS, total transmitted and received data, packet loss, and throughput providing techniques are run and assessed and the simulation results are examined. According to the results, appropriate QoS adaption allows for specific voice and video transmission. Finally, by analyzing existing QoS parameters, the QoS performance of 4G/UMTS networks may be improved.


2021 ◽  
Vol 42 (9) ◽  
pp. 092301
Author(s):  
Yangyang Deng ◽  
Yuehui Wang ◽  
Yiqing Zhang ◽  
Axin Du ◽  
Jianguo Liu

2021 ◽  
Vol 1335-8243 (1338-3957) ◽  
pp. 30-34
Author(s):  
Cyril Filip KOVACIK ◽  
◽  
Gabriel BUGAR ◽  

Voice transmission over the Internet network is now taken for granted. Many end-user applications address this issue. However, this paper focuses on the specific use of the SCCP protocol created by Cisco, its implementation in a computer network and end devices, determination of the operational properties of this implementation, and their comparison in different conditions. VoIP traffic is compared at different bandwidths and implemented by different configurations of IP protocols. By investigated implementations of IP protocols are meant IPv4, IPv6, and IPv4 protocol with applied NAT. As part of the application of various IP protocols is also compared VoIP communication with a video stream on a local basis. The conclusion of the paper is devoted to the graphical evaluation of these observations and to draw conclusions based on them.


Author(s):  
K. Durga Sowjanya ◽  
P. Bhaskara Reddy

Everyone necessitate communication technology in this information era. By use of modern technology, everyone should prefer wireless service without using bundle of wires. Colleges, shopping malls, corporate offices, coffee shops, construction buildings and restaurants provide Wi-Fi service to customers. Wi-Fi service allows data and voice transmissions over connected devices in same network. Proposed paper presents development of voice call transfer service that processes incoming telephone call which uses Wi-Fi as communication medium for voice transmission between smart phone and tablet. The proposed method allows real time communication transferred over wifi using IP between smart phone and tablet at no cost. Wi-Fi enabled smart phone and tablet are connected to the router or wifi access point to provide communication between caller and tablet user. Proposed paper develops cost effective, easy and reliable voice communication over Wi-Fi which provides good and comfort experience to user.


Author(s):  
Il Yeong Cho ◽  
Ji Yeon Lee ◽  
Tae Geon Park ◽  
Kyung Bae Kim ◽  
Yeon Man Jeung ◽  
...  

2021 ◽  
Vol 11 (4) ◽  
pp. 1818
Author(s):  
Chin-Feng Lin

An underwater universal filtered multicarrier (UFMC)-based voice transmission scheme is proposed using a 512-point inverse discrete Fourier transform, utilizing 10 sub-bands, and that each had 20 subcarriers. In this proposed UFMC method, the adaptive modulation technologies with 4 quadrature amplitude modulation (QAM), 16-QAM, and low-density parity-check (LDPC) channel coding were integrated. Additionally, the bit error rate (BER), transmission power weighting, the ratios of power-saving, and underwater voice transmission performance with perfect channel estimation (PCE), and 5% and 10% channel estimation errors (CEEs) were investigated. The underwater voice transmission had a BER quality of service 10−3. Simulation results showed that the PCE outperformed 5% and 10% CEEs, under 4-QAM, with gains of 0.5 and 0.9 dB, respectively, and a BER of 4×10−4. The PCE outperformed 5% and 10% CEEs, under 16-QAM, with gains of 0.5 and 2.4 dB, respectively, and a BER of 8.5×10−4. The proposed UFMC scheme can be applied to underwater voice transmission with a BER below 10−3 The proposed system showed a superior capability to contend with additive white Gaussian noise, underwater multipath channel fading, and CEEs.


2021 ◽  
Vol 75 (2) ◽  
pp. 175-181
Author(s):  
Maksym Smyrnov ◽  

The article deals with specific controversial issues concerning features of using videoconferencing in criminal proceedings in general as well as at its specific stages; also own conclusions and propositions are justified which are aimed at further development of criminal procedural legislation of Ukraine with the regard to the questions raised. The essence, meaning, advantages, current state, and perspectives of using videoconferencing in criminal proceedings of Ukraine and the area of international cooperation among the states in criminal justice are examined. The state of legal regulation of using videoconferencing in criminal proceedings is analyzed. Code of Criminal Procedure of Ukraine provides for using videoconferencing both in criminal proceedings of Ukraine and criminal proceedings with a foreign element. Based on the analysis of grounds and procedure of video-conferencing features of its usage both in criminal proceedings of Ukraine and in the area of international cooperation among the states in criminal justice are identified. Videoconferencing can be used based on the decision of an investigator, prosecutor, investigative judge, or court in each case taking into account circumstances of criminal proceedings and subject to grounds provided for by the Code of Criminal Procedure of Ukraine. Before the start of investigative (search) action or court session using video-conferencing one shall ensure that nothing prevents a person from giving testimony, making motions, providing evidence, etc. This fact is essential for the admissibility of evidence used in criminal proceedings obtained using videoconferencing. Conducting requested procedural actions by video-conferencing ensure that an accused person, a victim, and other participants have an opportunity to express themselves on the raised issues, make arguments aimed to rebut the conclusions of the opposing party, provide evidence and make motions during pre-trial investigation or trial. Specificities of conducting a questioning using videoconferencing in the area of international cooperation as well as advantages of obtaining testimony from individuals put into custody or serving a sentence in a foreign state using videoconferencing compared to traditional means are formulated. The issue of interrelation between videoconferencing and principles of criminal proceedings is examined and it is showed that using videoconferencing is fully consistent with principles of criminal proceedings. Videoconferencing is one of the procedural forms of using information technologies in criminal proceedings and is used to conduct an action, participants of which are geographically separated one from another and thus communication among them are conducted using the communication technologies that support real-time image and voice transmission.


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