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1
KOVACIK, Cyril Filip, and Gabriel BUGAR. "ANALYSIS OF OPERATIONAL PROPERTIES OF VOIP NETWORK." Acta Electrotechnica et Informatica 1335-8243, no. 1338-3957 (2021): 30–34. http://dx.doi.org/10.15546/aeei-2021-0005.
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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.
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Prijono, Wahyu Adi. "PENGARUH PENGGUNAAN CODEC STANDART ITU G.729 TERHADAP SISTEM KOMUNIKASI VOIP." SISTEM Jurnal Ilmu Ilmu Teknik 17, no. 1 (2021): 11–22. http://dx.doi.org/10.37303/sistem.v17i1.193.
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Voice over Internet Protocol (VoIP) is a technology that is capable of passing voice traffic, in the form of packets through the network Internet Protocol (IP). IP network itself is a data communications network based packet-switch. The voice signal before experiencing bundled voice coding or format conversion of sound into digital form that can be passed over an IP network. Telephony, Internet telephony, or termed VoIP (Voice Over Internet Protocol.This communication system use VoIP (Voice over Internet Protocol), ie voice calls over data services (internet). This communication was developed using Android-based devices. Based on characteristics, android devices are open source, so users do not need to have a license to be able to have android-based devices. In addition, the android device that must be connected to a SIP (Session Iniation Protocol) is a data service that can be done with a paid subscription of the user of the operator using a conventional pulse. Telecommunications designed will use a hybrid system, the merger between VoIP communications with data communications GSM network. With basic calculations where Coding standards G 729, is a standard that can be used for voice communication system through data networks with rate of 8 Kbps. The implementation of the G729 codec is effect on communication systems VOIP.
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Daramola, Oladunni Abosede. "QUALITY OF SERVICE ISSUES IN WIRELESS VOICE OVER INTERNET PROTOCOL." International Journal of Advanced Research in Computer Science and Software Engineering 7, no. 10 (2017): 57. http://dx.doi.org/10.23956/ijarcsse.v7i10.386.
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Voice over Internet Protocol (VoIP) is a significant application of the converged network principle where the voice traffic is routed over Internet Protocol shared traffic networks. VoIP traffic was modelled over wireless network and a simulation of the traffic was transmitted over the network. E-model technique was used to analyze the traffic data and also to rate VoIP QoS parameters. The result achieved was mapped to the Mean Opinion Scale to determine the Quality of Service of VoIP over wireless networks. The results shows that QoS in the VoIP communications is significantly impacted by these parameters and the impact varies according to the parameters and also the communication aspects selected for the VoIP traffic analysis.Keywords: VoIP, QoS, E-Model and Mean Opinion Scale
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Arif, Rabbai San, Yuli Fitrisia, and Agus Urip Ari Wibowo. "Implementasi Voip Server Berbasis IPV6 Dengan Raspberry PI." Manutech : Jurnal Teknologi Manufaktur 9, no. 01 (2019): 47–54. http://dx.doi.org/10.33504/manutech.v9i01.32.
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Voice over Internet Protocol (VoIP) is a telecommunications technology that is able to pass the communication service in Internet Protocol networks so as to allow communicating between users in an IP network. However VoIP technology still has weakness in the Quality of Service (QoS). VOPI weaknesses is affected by the selection of the physical servers used. In this research, VoIP is configured on Linux operating system with Asterisk as VoIP application server and integrated on a Raspberry Pi by using wired and wireless network as the transmission medium. Because of depletion of IPv4 capacity that can be used on the network, it needs to be applied to VoIP system using the IPv6 network protocol with supports devices. The test results by using a wired transmission medium that has obtained are the average delay is 117.851 ms, jitter is 5.796 ms, packet loss is 0.38%, throughput is 962.861 kbps, 8.33% of CPU usage and 59.33% of memory usage. The analysis shows that the wired transmission media is better than the wireless transmission media and wireless-wired.
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Handayani, Tri Febriana, Pande Ketut Sudiarta, and I. Made Oka Widyantara. "UJI KEAMANAN KOMUNIKASI VOIP MENGGUNAKAN SISTEM KEAMANAN SRTP-TLS PADA JARINGAN NIRKABEL." Jurnal SPEKTRUM 5, no. 1 (2018): 13. http://dx.doi.org/10.24843/spektrum.2018.v05.i01.p02.
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VoIP is a technology used to communicate voice remotely and directly using data communication lines with TCP / IP protocol. But basically, VoIP communication does not guarantee data security when doing communication. A very important security system is added in VoIP communications to maintain the confidentiality of communication, so that communication can not be recorded and played back. To build a security when communicating VoIP, then in this study added an SRTP-TLS security system. The study was conducted on wireless networks by comparing the security of data communications when using the SRTP-TLS security system and without using a security system.Keywords: Wireless, Security System, VoIP
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Riki, Riki, Aditiya Hermawan, and Yusuf Kurnia. "Voice Over Internet Protocol Based Communication Design (VoIP) With 3CXSystemPhone On Android Smartphone." bit-Tech 1, no. 1 (2018): 1–8. http://dx.doi.org/10.32877/bt.v1i1.2.
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TCP\IP protocol can be connected to various computer data networks in the world. This protocol increasingly exists and is needed so that many parties develop it to vote through this protocol. Voice Over Internet Protocol technology is the answer to that desire. This technology is able to convert analog voice (human voice) into data packets then through public internet data networks and private intranet data packets are passed, so that communication can occur. With VoIP communication costs can be reduced so that it can reduce investment costs and conversations (cost saving) or even up to 100% free. VoIP implementation can be done by designing a wireless VoIP network (cable) using 3CXSystemPhone software as a PBX. In this scientific work the software used is 3CXSystemPhone 11.0, where SIP is a VoIP server which is a freeware software, in its application only requires one PC server and several PC clients (2 for example) that are connected to each other
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Pasarelski, Rosen, and Verginya Todorova. "Analysis of protocols and techniques for transmission of voice over internet protocol." Yearbook Telecommunications 6 (September 29, 2019): 105–13. http://dx.doi.org/10.33919/ytelecomm.19.6.11.
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The purpose of the article is to present the process of the evolution of telecommunication networks across the years and are developing at a very rapid pace, with a tendency towards convergence of services. Voice over IP is the preferred method for more and more telecommunications operators, replacing standard telephony and networking. As a result, it can be noted that VoIP technology allows much more information to be transmitted over the network to serve and improve communication needs than traditional telephony. The authors' contribution is the analysis of voice over IP protocols, which clarifies the concepts and rules in this type of communication and presents the functionalities and components of these communications.
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Sudiarta, Pande Ketut, and I. Putu Ardana. "Implementation of Hotspot Network for Internal Campus Communications Utilizing Smartphone and Free Software." Journal of Electrical, Electronics and Informatics 1, no. 1 (2017): 33. http://dx.doi.org/10.24843/jeei.2017.v01.i01.p07.
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VoIP (voice over internet protocol) telephone communication using a data network. There is a change in switching from circuit switching technology into packet switching. Phone exchange can now use the Personal Computer equipped VoIP applications. Even the development of mobile phone technology to make VoIP communication can be performed utilizing the Smartphone. VoIP applications commonly use such as WhatsApp, Line etc. However, this application will cut the user data packets and often the quality is not satisfactory due to limited bandwidth and location of the remote server. During this time, Udayana University campus at several locations has been equipped with Hotspot network is mostly used to connect to the Internet. Hotspot network can also be used for voice communications with VoIP technology by adding a VoIP server. This concept not raises communication costs and should produce sound quality will be better because of the close location of the server. Because that researchers need to develop a model of telephone communication network utilizing Smartphone hotspot and students to be able to communicate in a campus environment. Method of this research is to develop a network utilizing hotspots and VoIP telephone exchange using the mini PC installed software FreePBX. On the side of the Smartphone using the free soft phone application and for aircraft used FXS analog phone as a codec. Tests will be performed for the communication between Smartphone and Smartphone to FXS terms of QOS and MOS produced. The results obtained, if the latency and packet loss have a value corresponding to the Real Time Protocol (RTP), the obtained MOS appropriate for the codec used while with the same codec if the value of packet loss and latency results are high then MOS obtained be small or less good quality. So the quality of VoIP is highly dependent on the quality of the signal obtained hotspot. In general, VoIP communication using a Smartphone connected to the server on the network hotspot mini pc can be used as a voice communication on campus.
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Honni, Honni. "Rancang Bangun Perangkat Lunak Billing dan Implementasi Voice Over Internet Protocol." ComTech: Computer, Mathematics and Engineering Applications 4, no. 2 (2013): 603. http://dx.doi.org/10.21512/comtech.v4i2.2483.
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The rapidly evolving communication system enables applications for telephone communication to be carried over the data network known as VoIP (voice over internet protocol). SIP (session initiation protocol) as the signaling protocol is text-based VoIP which can be implemented easily in comparison with other signalingprotocols. The purpose of this paper is designing and implementing VoIP billing up to the company to provide additional facilities for enterprise customers. The methods start with data collection, analysis, design, development, and implementation. The result achieved is a system of VoIP with SIP and Asterisk software which has functions of PBX to provide additional facilities such as VoIP which is a plus for the company and customers. After implemented, the VoIP system and billing features are found work well.
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Hirzan, Alauddin Maulana, Nazrulazhar Bahaman, and Whisnumurti Adhiwibowo. "Voice Over Internet Protocol Performance Evaluation in 6to4 Tunneling Network." Jurnal Transformatika 18, no. 1 (2020): 108. http://dx.doi.org/10.26623/transformatika.v18i1.2356.
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<span lang="EN-US">Registry reported that their regional already in exhausted state. The IPv6 was proposed to substitute IPv4 network, but the implementation of this version cased many problems such as hardware compatibility. As temporary solution to this problem, 6to4 tunneling transition mechanism is introduced as one of many solutions. This mechanism used IPv4 network as communication media between two IPv6 networks. Thus, this kind of mechanism will affect the performance of Voice over Internet Protocol. VoIP demanded real-time communication by using UDP protocol between nodes. Unlike normal communication mode, real-time mode required data to be sent immediately ignoring the quality of data. This research evaluated the performance of 6to4 tunneling mechanism for Voice over Internet Protocol’s communication between two nodes in native IPv6 networks. </span>
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Susiani Pande, Putu Sintia, Pande Ketut Sudiarta, and I. Made Oka Widyantara. "PENGUKURAN KINERJA VOIP DENGAN CODEC G.711?, G.711a DAN G.729 DI MEDIA TRANSMISI NIRKABEL BERBASIS SIP DAN IAX." Jurnal SPEKTRUM 5, no. 1 (2018): 21. http://dx.doi.org/10.24843/spektrum.2018.v05.i01.p04.
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Voice over Internet Protocol (VoIP) is a technology that can send real-time data with IP-based networks (Internet Protocol). In VoIP technology with wireless network has several problems that cause the performance of the network to be varied due to the QoS (Quality of Service) include delay, jitter, packet loss and MOS that affect the wireless network. This research uses G.711?, G.711a and G.729 codec based on SIP and IAX server on wireless network which then the QoS result from each codec compared with ITU-T standard which become the reference of whether the network is good or not so that later can realized on campus. In the research results, QoS on wireless IEEE 802.11 b has linear results, whereas QoS wireless in VoIP has fluctuating results because the use of codecs in VoIP on each codec has a large bitrate and different coding techniques and is a feature of wireless networks. The QoS comparison of three codecs produced the best G711 Q7S codecs because the G.711 codec has a bitrate that conforms to the 64 kbps voice communication standard and uses voice coding techniques that match the digital signal encoding technique of PCM (Pulse Code Modulation).
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Luhach, Ravindra, Chandra K. Jha, and Ashish K. Luhach. "Performance Analysis of QMF Filter Bank For Wireless Voip in Pervasive Environment." Recent Patents on Computer Science 12, no. 4 (2019): 349–53. http://dx.doi.org/10.2174/2213275911666181018101737.
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Background: Voice over Internet Protocol (VoIP) has emerged as one of the most significant technology in the field of communication and evolved as a substitute to the conventional communication method as the Public Switched Telephone Network (PSTN). Along with the advantages such as scalability and security, VoIP has some threats such as voice quality and interference that must be dealt with. The voice quality in VoIP is degraded when transmitted over a computer network due to delay, jitter and packet loss etc. Packet loss is one of major reasons for the signal quality degradation. Objective: In this research article, Quadrature Mirror Filter Bank (QMF) has been implemented in wireless VoIP system to enhance the quality of the signals transmitted. Results: The performance has been evaluated under varying network conditions of packet loss. Conclusion: Significant improvement has been observed in the quality of VoIP signal.
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Leu, Jenq-Shiou, Wei-Hsiang Lin, Wen-Bin Hsieh, and Chien-Chih Lo. "Design and Implementation of a VoIP Broadcasting Service over Embedded Systems in a Heterogeneous Network Environment." Scientific World Journal 2014 (2014): 1–10. http://dx.doi.org/10.1155/2014/917060.
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As the digitization is integrated into daily life, media including video and audio are heavily transferred over the Internet nowadays. Voice-over-Internet Protocol (VoIP), the most popular and mature technology, becomes the focus attracting many researches and investments. However, most of the existing studies focused on a one-to-one communication model in a homogeneous network, instead of one-to-many broadcasting model among diverse embedded devices in a heterogeneous network. In this paper, we present the implementation of a VoIP broadcasting service on the open source—Linphone—in a heterogeneous network environment, including WiFi, 3G, and LAN networks. The proposed system featuring VoIP broadcasting over heterogeneous networks can be integrated with heterogeneous agile devices, such as embedded devices or mobile phones. VoIP broadcasting over heterogeneous networks can be integrated into modern smartphones or other embedded devices; thus when users run in a traditional AM/FM signal unreachable area, they still can receive the broadcast voice through the IP network. Also, comprehensive evaluations are conducted to verify the effectiveness of the proposed implementation.
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Armanto, Armanto Armanto. "IMPLEMENTASI SERVER VoIP (Voice Over Internet Protocol ) PADA KANTOR KECAMATAN SALING KABUPATEN EMPAT LAWANG." Jurnal Sistem Komputer Musirawas (JUSIKOM) 3, no. 2 (2018): 114. http://dx.doi.org/10.32767/jusikom.v3i2.344.
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Implementation of VoIP Server At Sub-District Office of Saling Regency Empat Lawang has building structure which has many room. In case communicate sometimes takes a fast time. Like a meeting to be made at that moment. For that to be helpful in time and efficient then it can be applied Conference Call at Saling sub-district office. In this study the network used is Local Area Network (LAN), Using server3CX Phone System, while in developing the system used NDLC (Network Development Life Cycle) method. With VoIP Server Implementation In Sub-District Office of Saling Regency Empat Lawang can make inter-space communication to be efficient and reduce the cost of usage of mobile phone pulse which is too high. Keywords: VoIP server, 3CX Phone System, Conference
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Vakilinia, Shahin, Mohammadhossein Alvandi, Mohammadreza Khalili Shoja, and Iman Vakilinia. "Cross-Layered Secure and QoS Aware Design of VOIP over Wireless Ad-Hoc Networks." International Journal of Business Data Communications and Networking 9, no. 4 (2013): 23–45. http://dx.doi.org/10.4018/ijbdcn.2013100102.
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In this paper, Cross-layer design has been used to provide quality of service (QoS) and security at the same time for VOIP over the wireless ad-hoc network. In this paper the authors extend their previous work (i.e. Multi-path Multi-Channel Protocol Design for Secure QoS-Aware VOIP in Wireless Ad-Hoc Networks) by adding transport and application layers considerations. The goal of this paper is to support QoS and security of VOIP simultaneously. Simulation results shows that the proposed cross-layered protocol stack design significantly improve QoS parameters of the VOIP calls under the jamming or Denial-of-service attacks.
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Nisar, Kashif. "Voice Priority Queue Scheduling System Models for VoIP over WLANs." International Journal of Information Communication Technologies and Human Development 5, no. 1 (2013): 36–59. http://dx.doi.org/10.4018/jicthd.2013010103.
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The Voice over Internet Protocol (VoIP) is a delay sensitive traffic due to real-time applications on networks. The assessment of voice flow quality in the VoIP is an essential requirement for technical and commercial motivation. The packets of VoIP streaming may experience drops because of the competition among the different kinds of traffic flow over the network. A VoIP application is also sensitive to delay and requires the voice packets to arrive on time from the sender to the receiver side without any delay over WLAN. The scheduling system model for VoIP traffic is an unresolved problem. In this research paper, the author proposes a new Voice Priority Queue (VPQ) scheduling system models and algorithms for the VoIP over WLANs to solve scheduling issues over IP-based networks. They present new contributions, through the three stages of the VPQ. The VPQ scheduling algorithm is provided as an essential technique in the VoIP communication networks to guarantee the QoS requirements. The design of the VPQ is managed by the limited bandwidth utilization and has been proven to have an efficient performance over WLANs.
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Atmadja, Martono Dwi. "Single Board Computer Applications as Multi-Server VoIP." International Journal for Research in Applied Science and Engineering Technology 9, no. VII (2021): 1023–28. http://dx.doi.org/10.22214/ijraset.2021.36512.
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Telecommunication technology is developing along with information technology and several innovations in several audio and data transmission and reception techniques. Innovation and communication technology are hoped to be able to create efficiencies in regards to time, equipment, and cost. The Public Switched Telephone Network (PSTN) telephone technology has experienced integration towards communication using Internet Protocol (IP) networks, better known as Voice over Internet Protocol (VoIP). VoIP Technology transmits conversations digitally through IP-based networks, such as internet networks, Wide Area Networks (WAN), and Local Area Networks (LAN). However, the VoIP cannot fully replace PSTN due to several weaknesses, such as delay, jitter, packet loss, as well as security and echo. Telephones calls using VoIP technology are executed using terminals in the form of computer devices or existing analogue telephones. The benefit of VoIP is that it can be set in all ethernet and IP addresses. Prefixes can be applied for inter-server placements as inter-building telephone networks without the addition of inefficient new cables on single board computers with Elastix installed. Prefix and non-prefix analysis on servers from single board computers can be tested using QoS for bandwidth, jitter, and packet loss codec. The installation of 6 clients, or 3 simultaneous calls resulted in a packet loss value in the prefix Speex codex of 2.34%. The bandwidth in the prefix PCMU codec has an average value of 82.3Kbps, and a non-prefix value of 79.3Kbps, in accordance to the codec standards in the VoIP. The lowest jitter was found in the non-prefix PCMU codec with an average of 51.05ms, with the highest jitter for the prefix Speex codec being 314.65ms.
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Wu, Zhijun, Junjun Guo, Chenlei Zhang, and Changliang Li. "Steganography and Steganalysis in Voice over IP: A Review." Sensors 21, no. 4 (2021): 1032. http://dx.doi.org/10.3390/s21041032.
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The rapid advance and popularization of VoIP (Voice over IP) has also brought security issues. VoIP-based secure voice communication has two sides: first, for legitimate users, the secret voice can be embedded in the carrier and transmitted safely in the channel to prevent privacy leakage and ensure data security; second, for illegal users, the use of VoIP Voice communication hides and transmits illegal information, leading to security incidents. Therefore, in recent years, steganography and steganography analysis based on VoIP have gradually become research hotspots in the field of information security. Steganography and steganalysis based on VoIP can be divided into two categories, depending on where the secret information is embedded: steganography and steganalysis based on voice payload or protocol. The former mainly regards voice payload as the carrier, and steganography or steganalysis is performed with respect to the payload. It can be subdivided into steganography and steganalysis based on FBC (fixed codebook), LPC (linear prediction coefficient), and ACB (adaptive codebook). The latter uses various protocols as the carrier and performs steganography or steganalysis with respect to some fields of the protocol header and the timing of the voice packet. It can be divided into steganography and steganalysis based on the network layer, the transport layer, and the application layer. Recent research results of steganography and steganalysis based on protocol and voice payload are classified in this paper, and the paper also summarizes their characteristics, advantages, and disadvantages. The development direction of future research is analyzed. Therefore, this research can provide good help and guidance for researchers in related fields.
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Militani, Davi Ribeiro, Hermes Pimenta de Moraes, Renata Lopes Rosa, Lunchakorn Wuttisittikulkij, Miguel Arjona Ramírez, and Demóstenes Zegarra Rodríguez. "Enhanced Routing Algorithm Based on Reinforcement Machine Learning—A Case of VoIP Service." Sensors 21, no. 2 (2021): 504. http://dx.doi.org/10.3390/s21020504.
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The routing algorithm is one of the main factors that directly impact on network performance. However, conventional routing algorithms do not consider the network data history, for instances, overloaded paths or equipment faults. It is expected that routing algorithms based on machine learning present advantages using that network data. Nevertheless, in a routing algorithm based on reinforcement learning (RL) technique, additional control message headers could be required. In this context, this research presents an enhanced routing protocol based on RL, named e-RLRP, in which the overhead is reduced. Specifically, a dynamic adjustment in the Hello message interval is implemented to compensate the overhead generated by the use of RL. Different network scenarios with variable number of nodes, routes, traffic flows and degree of mobility are implemented, in which network parameters, such as packet loss, delay, throughput and overhead are obtained. Additionally, a Voice-over-IP (VoIP) communication scenario is implemented, in which the E-model algorithm is used to predict the communication quality. For performance comparison, the OLSR, BATMAN and RLRP protocols are used. Experimental results show that the e-RLRP reduces network overhead compared to RLRP, and overcomes in most cases all of these protocols, considering both network parameters and VoIP quality.
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Jeyanthi, N., R. Thandeeswaran, and J. Vinithra. "Rqa based approach to detect and prevent ddos attacks in voip networks." Cybernetics and Information Technologies 14, no. 1 (2014): 11–24. http://dx.doi.org/10.2478/cait-2014-0002.
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Abstract Voice over Internet Protocol (VoIP) is a family of technologies for the transmission of voice over Internet. Voice is converted into digital signals and transmitted as data packets. The Session Initiation Protocol (SIP) is an IETF protocol for VoIP and other multimedia. SIP is an application layer protocol for creating, modifying and terminating sessions in VoIP communications. Since SIP is a more flexible and simple protocol, it is quite easy to add features to it. Distributed Denial of Service Attack (DDoS) floods the server with numerous requests from various hosts. Hence, the legitimate clients will not be able to get their intended services. A major concern in VoIP and almost in all network domains is availability rather than data consistency. Most of the surviving techniques could prevent VoIP network only after collision. This paper proposes a Recurrence Quantification based approach to detect and prevent VoIP from a DDoS attack. This model detects the attack at an earlier stage and also helps to prevent from further attacks. In addition, this techniques enables the efficient utilization of resources. QUALNET has been used to simulate the operation of the proposed technology.
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Susanto, Panca. "QoS measurement of VoIP codec usage on limited bandwidth network over UDP-based VPN." Telfor Journal 12, no. 1 (2020): 13–17. http://dx.doi.org/10.5937/telfor2001013s.
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The usage of internet protocol for voice communication is widely used and more efficient rather than an analog signal. However, there is no security guaranteed on IP-based voice communication. The voice payload can be easily tapped or even manipulated. In the case of improving the security aspect, communication quality should be also considered. VoIP requires sufficient bandwidth to get proper communication quality. The ITU-T released a standard unit of communication quality, known as Mean Opinion Score (MOS) which is made from the subjective judgments of some individuals. However, MOS method takes time and is expensive. In this research, we measure VoIP communication which is secured by using VPN and build a tool for analyzing the voice packet between communication peers. The tool has capabilities to measure delay, jitter, and packet loss. Since VoIP has a QoS standard by ITU-T, the usage of VPN for security purpose needs to be considered. The sound quality might be decreased due to the addition of header for tunneling method, as well as the additional delay when the encryption processing is carried out. We used 3 types of codec: a-Law, GSM, and iLBC which will be passed on 4 types of bandwidth (256, 128, 64, 32 kbps) through the UDP-based VPN that use 3 types of encryption method (3-DES, Blowfish, AES).
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Yulianto, Budi. "Analisis Korelasi Faktor Perilaku Konsumen terhadap Keputusan Penggunaan Teknologi Komunikasi Voip." ComTech: Computer, Mathematics and Engineering Applications 5, no. 1 (2014): 236. http://dx.doi.org/10.21512/comtech.v5i1.2619.
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The advancement of communication technology that is combined with computer and the Internet brings Internet Telephony or VoIP (Voice over Internet Protocol). Through VoIP technology, the cost of telecommunications in particular for international direct dialing (IDD) can be reduced. This research analyzes the growth rate of VoIP users, the correlation of the consumer behavior towards using VoIP, and cost comparisons of using telecommunication services between VoIP and other operators. This research is using descriptive analysis method to describe researched object through sampling data collection for hypothesis testing. This research will lead to the conclusion that the use of VoIP for international area will be more advantageous than the use of other operators of GSM (Global System for Mobile), CDMA (Code Division Multiple Access), or the PSTN (Public Switched Telephone Network).
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Aminah, Nina Siti, Muhamamad Reza Ramadhani Raharjo, and Maman Budiman. "Low-cost wireless mesh communications based on openWRT and voice over internet protocol." International Journal of Electrical and Computer Engineering (IJECE) 11, no. 6 (2021): 5119. http://dx.doi.org/10.11591/ijece.v11i6.pp5119-5126.
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Technology makes it easier for us to communicate over a distance. However, there are still many remote areas that find it difficult to communicate. This is due to the fact that communication infrastructure in some areas is expensive to build while the profit will be low. This paper proposes to combine voice over internet protocol (VoIP) over mesh network implemented on openWRT router. The routers are performing mesh functions. We set up a VoIP server on a router and enabled session initiation protocol (SIP) clients on other routers. Therefore, we only need routers as a means of communication. The experiment showed very good results, in the line-of-sight (LOS) condition, they are limited to reception distances up to 145 meters while in the non-line-of-sight (NLOS) condition, they are limited to reception distances up to 55 meters.
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Supendar, Hendra, Yopi Handrianto, and Santoso Setiawan. "Kualitas Pelayanan Dalam Voice Over Internet Protokol Berbasis Shorewall." PIKSEL : Penelitian Ilmu Komputer Sistem Embedded and Logic 7, no. 2 (2019): 123–32. http://dx.doi.org/10.33558/piksel.v7i2.1815.
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Abstract
 
 Communication is very important and a success factor of the system at the company. Communication can be done using telephone media and internet media. PT. Interdev Prakarsa has several branches and communication between branches is still done using telephone media, and it is very costly for the company especially since the company is already using internet media in the network. The solution to this problem in this study was designing an internet-based technology as communication between branches, the technology is called Voice over Internet Protocol (VoIP) with an open source operating system and an open source firewall that was Shorewall. Result showed that after installation and testing, this firewall reliable enough to overcome the problem of attack problems from outside. The results of VoIP tightening on service quality found that the application of VoIP did not consume a lot of CPU Benchmarks, i.e. only 0.80 to 1.35 per cent. The bandwidth used is also very small between 86 to 86.8 kbps for incoming calls and 83.4 up to 84.3 kbps for outgoing calls. Communication built between VoIP peripherals has also been tested to run well because the value of delay, jitter and packet loss is included in the good category.
 Keywords: VoIP, Shorewall, Comunication
 
 Abstrak
 
 Komunikasi pada sebuah perusahaan sangatlah penting, dimana komunikasi menjadi alat ukur keberhasilan sistem di perusahaan tersebut. Komunikasi yang dilakukan dapat menggunakan media telepon dan media internet. PT. Interdev Prakarsa memiliki beberapa cabang dan komunikasi antar cabang masih dilakukan dengan menggunakan media telepon, dan itu sangatlah menghabiskan cost perusahaan apalagi perusahaan tersebut sudah menggunakan media internet dalam jaringan. Solusi dari permasalahan tersebut adalah dengan mendesain sebuah teknologi yang berbasis jaringan internet sebagai komunikasi antar cabang, teknologi tersebut bernama Voice over Internet Protocol (VoIP) dengan sebuah sistem operasi open source dan sebuah firewall open source yaitu Shorewall dimana setelah dilakukan penginstallan dan pengetesan, firewall ini cukup handal untuk mengatasi masalah masalah serangan dari luar. Hasil dari pengetasan VoIP terhadap kualitas pelayanan didapatkan bahwa penerapan VoIP tidak banyak menghabiskan Benchmark CPU yaitu hanya 0,80 sampai degan 1.35 persen per call. Untuk bandwidth yang di pakai juga sangatlah kecil berada di antara 86 sampai dengan 86.8 kbps untuk panggilan masuk dan 83.4 sampai dengan 84.3 kbps untuk panggilan keluar. Komunikasi yang dibangun antar peripheral VoIP pun teruji berjalan dengan baik karena nilai dari delay, jitter dan packet loss termasuk dalam katagori baik.
 
 Kata kunci: VoIP, Shorewall, Komunikasi
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Octavia, Hadria. "UNJUK KERJA PENERAPAN TEKNOLOGI VoIP PADA JARINGAN VPN (VIRTUAL PRIVATE NETWORK)." Elektron : Jurnal Ilmiah 5, no. 2 (2018): 1–12. http://dx.doi.org/10.30630/eji.5.2.49.
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VoIP ( Voice over Internet Protocol ) is a technology used for communication in the form of IP based voice media over long distances. The concept of a VPN (Virtual Private Network) in this paper makes a client that is on the public network can be connected to a LAN network. To use the VoIP server in the Linux operating system Trixbox, whereas for the VPN server using ClearOS and X-lite is used as a softphone to make calls to the client. Of testing at 64kbps bandwidth using the G711 codec produces value performance (delay, jitter, and packet loss ) is not good, so that voice data delivered is less clear. Thus the choice of bandwidth for the G.711 codec 512kbps up is the best solution to get the value of the performance (delay, jitter, and packet loss) better . And a choice of 3 Greed (low, medium, high) on setting bandwidth, high is the best option. Because it can produce the best performance for VoIP VPN technology.
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TAN, YUNG HAN, ARUN KRISHNAN THAMPI, DALEY JOSEPH SEBASTIAN, and YAJUN HA. "DESIGN OF SEAMLESS PROTOCOL SWITCHING LAYER FOR VOICE OVER INTERNET PROTOCOL (VOIP) THAT SWITCHES BETWEEN BLUETOOTH AND IEEE 802.11." International Journal of Software Engineering and Knowledge Engineering 15, no. 02 (2005): 271–78. http://dx.doi.org/10.1142/s0218194005002178.
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The capability of seamlessly switching between two communication protocols will be very important for communication devices of the future, since it allows the end users to judiciously use whichever network is appropriate, depending on cost, signal strength or other factors such as the amount of battery life left on the device. This paper presents the groundbreaking idea of a Seamless Protocol Switching Layer (SPSL) on a hardware and software level to solve this problem. In addition, the SPSL concept is implemented by developing a prototype application, a Smart Video Phone, built using Intel XScale-based PXA255 board and ARM Linux as the operating system that can seamlessly switch between IEEE 802.11 and Bluetooth technologies. Experiments show that if the signal of the Bluetooth signal goes below 40%, the switching to Wireless-Fidelity (Wi-Fi) happens if it is available.
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Widhiatmoko, Widhiatmoko, Hesti Susilawati, and Rahmat K. Noviandono. "Analisis Performansi VOIP (Voice Over Internet Protocol) Pada Jaringan Wimax (Worldwide Interoperability For Microwave Access) Di Wilayah DKI Jakarta." JURNAL INFOTEL - Informatika Telekomunikasi Elektronika 3, no. 1 (2011): 58. http://dx.doi.org/10.20895/infotel.v3i1.88.
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VoIP is a system that uses the Internet network to transmit voice packets from one place to another using IP protocols intermediaries. With VoIP technology can be much cheaper call charges, especially for communicating overseas because of voice and data using the same network ie the Internet network. VoIP is a service that is very susceptible to delay while the existing access network is currently providing a significant delay for VoIP. One alternative network that can be used to overcome these problems is to use WiMAX technology because WiMAX can provide speed data services up to 70 Mbps.From the research, results of one way delay, jitter and packet loss still at the value recommended by ITU-T, which is the maximum value of one way delay measurement is 159.87 ms, for jitter 7.52 ms and for packet loss is 3.175%. The one way delay and packet loss from the measurement used to find the MOS score which is the value for quality of VoIP. MOS value range obtained from the calculation of 3.6 to 4.2, which means VoIP feasible to apply to the WiMAX network. The maximum value can reach 2.8 Mbps throughput to 0.575 Mbps for downlink and uplink. From the research also found that the SQI values that are above the standard value of the device will provide a high SNR value, and the higher SQI values then its RSSI value is also bigger.
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Murhaban, Murhaban, and Ahmad Ashari. "Analisa Metode Handover Pada Jaringan WiMAX." IJCCS (Indonesian Journal of Computing and Cybernetics Systems) 10, no. 1 (2016): 59. http://dx.doi.org/10.22146/ijccs.11189.
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Handover method is used to keep the stabilization of connection. Its connected with the performance was caused the process canal traffic transfer automatically in mobile station (MS) that was used to communicate without cutting off the connection. The main factor of success in handover was quality of service to provide the difference level of service in arranging and giving the traffic priority in the network like voice over IP (VoIP) application or communication voice using internet network. This research will analyse the achievement quality of service in the WiMax network standard 802.16e used hard handover and softhandover method with the VoIP application in mobile station. Based on the testing that was carried out hard handover and soft handover method used the application of voice over internet protocol in mobile station has obtained value jitter 0.001 Ms – 0.31 ms, and delay 10.5 ms 39 ms this is proved that the influence of jitter and delay against handover with the VoIP application still in the tolerance stage that was permitted. It is different with the output throughput 85 Bit/Sekon - 550 Bit/Sekon that is too low and indicated that throughput is not sentitif against handover with the voice over internet protocol application.
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Stojanovic, Mirjana D., and Vladanka S. Acimovic-Raspopovic. "Communication Issues for Small and Medium Enterprises." International Journal of Productivity Management and Assessment Technologies 1, no. 4 (2012): 41–61. http://dx.doi.org/10.4018/ijpmat.2012100103.
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This article considers communication issues for small and medium enterprises (SMEs) from both provider and customer perspectives. SME communication infrastructure at the individual site should usually be built around Ethernet-based local area network with a remotely manageable integrated access device that enables high speed Internet access, virtual private networking, Voice over Internet Protocol (VoIP) functionality and collaborative services. The authors further address several open quality of service (QoS) issues that include: service level agreements, signaling for quality of service and management aspects. The proposed framework for service management encompasses interfaces for QoS-aware and legacy applications, generic service level specification, functional model of service negotiation and management policies.
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Libnik, Rok, and Ales Svigelj. "Adaptive probe-based congestion-aware handover procedure using SIP protocol." International Journal of Computers Communications & Control 10, no. 5 (2015): 686. http://dx.doi.org/10.15837/ijccc.2015.5.1682.
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Wireless technologies have evolved very rapidly in recent years. In the future, operators will need to enable users to use communication services independently of access technologies, so they will have to support seamless handovers in heterogeneous networks. In this paper we present a novel adaptive congestion aware SIP based procedure for handover in heterogeneous networks. In the proposed algorithm the handover decision is based in addition to signal strength, also on target network congestion status, which is tested during the conversation. As SIP protocol was used, the proposed procedure is independent of access technologies. For performance evaluation of proposed procedure we developed a purpose‑built simulation model. The results show that the use of the proposed adaptive procedure significantly improves the QoE of VoIP users, compared to reference scenario, in which only signal strength was used as the trigger for handover decision.
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Li, Li Fu, Hong Liang Li, and Yong Jun Gu. "Research on Improving Strategies of PoC Session Setup Delay Based on TD-SCDMA Network." Applied Mechanics and Materials 701-702 (December 2014): 1000–1003. http://dx.doi.org/10.4028/www.scientific.net/amm.701-702.1000.
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PoC is a half-duplex PTT service using VoIP technology based on cellular network packet domain carrying. However PoC uses the standard Session Initiation Protocol (SIP) to setup session. The standard SIP signaling is too long, which brings more setup delay and does not hold true the wireless communication network, so it must be cut. This paper researchs the method to reduce PoC session setup delay from SIP session cut. We achieve the testing results in the actual commercial network. The results shows that this method can effectivly shorten the PoC session setup delay.
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ARYANTA, DWI, ARSYAD RAMADHAN DARLIS, and ARDHIANSYAH PRATAMA. "Implementasi Sistem IP PBX menggunakan Briker." ELKOMIKA: Jurnal Teknik Energi Elektrik, Teknik Telekomunikasi, & Teknik Elektronika 1, no. 2 (2013): 117. http://dx.doi.org/10.26760/elkomika.v1i2.117.
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ABSTRAKVoIP (Voice over Internet Protocol) adalah komunikasi suara jarak jauh yang digunakan melalui jaringan IP. Pada penelitian ini dirancang sistem IP PBX dengan menggunakan teknologi berbasis VoIP. IP PBX adalah perangkat switching komunikasi telepon dan data berbasis teknologi Internet Protocol (IP) yang mengendalikan ekstension telepon analog maupun ekstension IP Phone. Software VirtualBox digunakan dengan tujuan agar lebih memudahkan dalam sistem pengoperasian Linux yang dimana program untuk membuat IP PBX adalah menggunakan Briker yang bekerja pada Operating System Linux 2.6. Setelah proses penginstalan Briker pada Virtualbox dilakukan implementasi jaringan IP PBX. Setelah mengimplementasikan jaringan IP PBX sesuai dengan topologi, kemudian melakukan pengujian success call rate dan analisis Quality of Service (QoS). Pengukuran QoS menggunakan parameter jitter, delay, dan packet loss yang dihasilkan dalam sistem IP PBX ini. Nilai jitter sesama user Briker (baik pada smartphone maupun komputer) mempunyai rata-rata berada pada nilai 16,77 ms. Sedangkan nilai packetloss yang didapat pada saat terdapat pada saat user 1 sebagai pemanggil telepon adalah 0%. Sedangkan persentase packet loss pada saat user 1 sebagai penerima telepon adalah 0,01%. Nilai delay pada saat berkomunikasi antar user berada pada 11,75 ms. Secara keseluruhan nilai yang didapatkan melalui penelitian ini, dimana hasil pengujian parameter-parameter QOS sesuai dengan standar yang telah direkomendasikan oleh ITU dan didapatkan nilai QoS dengan hasil “baik”.Kata Kunci: Briker, VoIP, QoS, IP PBX, Smartphone.ABSTRACTVoIP (Voice over Internet Protocol) is a long-distance voice communications over IP networks are used. In this study, IP PBX systems designed using VoIP -based technologies. IP PBX is a telephone switching device and data communication technology-based Internet Protocol (IP) which controls the analog phone extensions and IP Phone extensions. VirtualBox software is used in order to make it easier for the Linux operating system to create a program which is using briker IP PBX that works on Linux 2.6 Operating System. After the installation process is done briker on Virtualbox IP PBX network implementation. After implementing the IP PBX network according to the topology, and then do a test call success rate and analysis of Quality of Service (QoS). Measurement of QoS parameters using jitter, delay, and packet loss resulting in the IP PBX system. Jitter value briker fellow users (either on a smartphone or computer) has been on the average value of 16.77 ms. While the values obtained packetloss when there is 1 user when a phone caller is 0%. While the percentage of packet loss at user 1 as a telephone receiver is 0.01%. Delay value when communicating between users located at 11.75 ms. Overall value obtained through this study , where the results of testing the QOS parameters in accordance with the standards recommended by the ITU and the QoS values obtained with the results "good".Keywords: Briker, VoIP, QoS, IP PBX, Smartphone.
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Mitra, Sulata, and Anup Roy. "Communication Void Free Routing Protocol in Wireless Sensor Network." Wireless Personal Communications 82, no. 4 (2015): 2567–81. http://dx.doi.org/10.1007/s11277-015-2365-7.
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De Rango, Floriano, Peppino Fazio, Francesco Scarcello, and Francesco Conte. "A New Distributed Application and Network Layer Protocol for VoIP in Mobile Ad Hoc Networks." IEEE Transactions on Mobile Computing 13, no. 10 (2014): 2185–98. http://dx.doi.org/10.1109/tmc.2014.2307315.
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Ali, Ali Mohd, Mahmoud Dhimish, Malek M. Alsmadi, and Peter Mather. "Algorithmic Identification of the Best WLAN Protocol and Network Architecture for Internet-Based Applications." Journal of Information & Knowledge Management 19, no. 01 (2020): 2040011. http://dx.doi.org/10.1142/s0219649220400110.
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This research developed a novel algorithm to evaluate internet-based services such as VoIP, Video Conferencing, HTTP and FTP, of different IEEE 802.11 technologies in order to identify the optimum network architecture among Basic Service Set (BSS), Extended Service Set (ESS) and the Independent Basic Service Set (IBSS). The proposed algorithm will yield the rank order of different IEEE 802.11 technologies. By selecting the optimum network architecture and technology, the best overall network performance that provides good voice, video and data quality is guaranteed. Furthermore, it meets the acceptance threshold values for the VoIP, Video Conferencing, HTTP and FTP quality metrics. This algorithm was applied to various room sizes ranging from [Formula: see text][Formula: see text]m to [Formula: see text][Formula: see text]m and the number of nodes ranged from 1 to 65. The spatial distributions considered were circular, uniform and random. The Quality of Service (QoS) metrics used were delay, jitter, throughput and packet loss.
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Oh, Hyung-Jun, Jae-Kyoung Park, and Yoo-Hun Won. "Redesign and Performance Analysis of RTP(Real-time Transport Protocol) for Encryption of VoIP Media Information between Different Communication Networks." Journal of the Korea Society of Computer and Information 18, no. 4 (2013): 87–96. http://dx.doi.org/10.9708/jksci.2013.18.4.087.
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Hemavathy, N., S. Sudha, and K. Ramesh. "A Dynamic Routing Path Reduction Protocol in Underwater Wireless Sensor Network." Sensor Letters 18, no. 5 (2020): 379–88. http://dx.doi.org/10.1166/sl.2020.4238.
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Recently, underwater wireless sensor networks (UWSNs) have emerged as a promising networking technique for various underwater applications. An energy efficient routing protocol plays a vital role in data transmission and practical applications. However, due to the specific characteristics of UWSNs, such as dynamic structure, narrow bandwidth, rapid energy consumption, and high latency, it is difficult to build routing protocols for UWSNs. In this paper, We propose a location aware opportunistic routing algorithm for under water communication. We analyse three main problems in under water communication; forwarding set selection forwarding set ranking to handle FSR problem, void handling method to handle the communication void (CV) and overhear and suppression procedure to deal with duplicate forwarding suppression (DFS) problems. The importance of the work is that it will provide an energy efficient pressure based opportunistic routing algorithm for wireless sensor network (UWSN). The routing protocol has been implemented in the ns2-AqaSim simulator and testbed for measurement of the performance metrics of the UASN. The simulation results showed that the novel routing method throughput has increased by 16%, 33%, and 55% when compared with SUN, VBF and DF method. It can effectively improve the throughput of nodes, balance positioning performance as well as energy use efficiency, and optimize the positioning result of UWASN.
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Ghoreyshi, Seyed Mohammad, Alireza Shahrabi, and Tuleen Boutaleb. "A Stateless Opportunistic Routing Protocol for Underwater Sensor Networks." Wireless Communications and Mobile Computing 2018 (November 11, 2018): 1–18. http://dx.doi.org/10.1155/2018/8237351.
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Routing packets in Underwater Sensor Networks (UWSNs) face different challenges, the most notable of which is perhaps how to deal with void communication areas. While this issue is not addressed in some underwater routing protocols, there exist some partially state-full protocols which can guarantee the delivery of packets using excessive communication overhead. However, there is no fully stateless underwater routing protocol, to the best of our knowledge, which can detect and bypass trapped nodes. A trapped node is a node which only leads packets to arrive finally at a void node. In this paper, we propose a Stateless Opportunistic Routing Protocol (SORP), in which the void and trapped nodes are locally detected in the different area of network topology to be excluded during the routing phase using a passive participation approach. SORP also uses a novel scheme to employ an adaptive forwarding area which can be resized and replaced according to the local density and placement of the candidate forwarding nodes to enhance the energy efficiency and reliability. We also make a theoretical analysis on the routing performance in case of considering the shadow zone and variable propagation delays. The results of our extensive simulation study indicate that SORP outperforms other protocols regarding the routing performance metrics.
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Kaur, Simranjeet, and Maninder Singh. "Intra-flow Contention Scheme for Improving QoS in WLAN." International Journal of Advanced Research in Computer Science and Software Engineering 7, no. 8 (2017): 106. http://dx.doi.org/10.23956/ijarcsse.v7i8.33.
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With the increasing popularity of wireless local area network(WLAN),the demand for multimedia services encompassing VoIP, multimedia streaming and interactive gaming is increasing rapidly. The real-time services require stringent Quality of Service (QoS) guarantees for effective communication. While a lot of research has dealt with providing QoS support for real-time services in traditional wired networks, the shared and broadcast nature of the wireless medium necessitates the design of new solutions for wireless networks. In wireless networks, unlike wired networks, the communication from one node will consume the bandwidth of the neighboring nodes and hence the shared bandwidth can be easily over-utilized. Therefore, to provide an acceptable level of QoS for the real-time services, it is necessary to control the utilization of the shared bandwidth. In this paper , we propose an efficient admission control scheme named Intra-flow contention scheme with CAC-OLSR routing protocol for WLAN networks which aims at preserving the QoS for all the admitted flows by employing a low overhead threshold mechanism. We describe several alternatives for the design of IAC and compare the performance of these alternatives using simulation results.
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Svrzić, Slađan, Zoran Miličević, and Zoran Perišić. "Description of the process of tunneling Q signaling in private telecommunication networks." Vojnotehnicki glasnik 69, no. 1 (2021): 31–63. http://dx.doi.org/10.5937/vojtehg69-28117.
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Introduction/purpose: The article should specify the network signaling type Q-SIG, which is standardized especially for implementation in digital telecommunication networks of integrated services (ISDN), emphasizing the possibility of its further application in the Private Telecommunications Network of Integrated Services of the Serbian Armed Forces (PISN of SAF), i.e. in the Private Automatic Telephone Network of the Serbian Armed Forces (PATN of SAF). Methods: An analysis of the existing standards was performed: ECMA355 and ECMA-336 and a synthesis of the possibilities of their application in the PATN of SAF. Results: The procedure for the application of Q-SIG is processed in a situation when the peripheral parts of the PISN of SAF, which operate on the principle of transmission and circuit switching by TDM (Time Division Multiplexing), are connected via a central Core network with the IP (Internet Protocol), which operates on the principle of packet transmission and switching with the SIP (Session Initiation Protocol). A method of the application of the tunneling of encapsulated Q-SIG messages through the IP network, defined by ECMA-355 Standard, has been developed. The necessary functions for mapping the transmission of tunneled signaling messages Q-SIG and mapping voice (and other audio) information to media streams during VoIP (Voice over IP) communication through that network, which are defined by ECMA-336 Standard, are described. Conclusion: The application of ECMA-355 and ECMA-336 Standards is a new solution in the PATN of SAF with the use of the IP network to connect the IP PINX using the Q-SIG tunneling procedures and mapping functions for their transmission and transmission of audio signals. This then opens up a whole range of new possibilities that, with the growth of the Core network and their application, will rapidly contribute to the creation of a broad Telecommunication information system backbone for the implementation of real-time multimedia communications and the transition to Unified Communications (UC).
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Shugran, Mahmoud Ali Al. "Non-Delay Tolerant Non-Overlay Routing Protocols Performance Evaluation for VANET." Computer and Information Science 14, no. 4 (2021): 20. http://dx.doi.org/10.5539/cis.v14n4p20.
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A Vehicular Ad hoc Network (VANET) is a distinctive situation of wireless ad hoc networks. The designing of the routing protocol considers a critical role in communication in VANET. VANET has specific features compared to other types of wireless ad hoc networks that impose special characteristics for designing of efficient routing protocols.The challenging factor in designing efficient routing protocols for VANET is the high movement of vehicles that incurs a rapid change in the network topology that causes frequent link breakage. This paper presents and evaluates different position-based routing protocols associated with VANETs. The evaluation aiming to determine appropriate specifications for optimal routing protocols&rsquo; features achieving best performance within different environmental conditions. The performance comparison is carried out in terms of Packet Delivery Rate (PDR), Void Problem Occurrence Rate (VPOR), and Average Hops Count (AHC).
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Tigarev, V., P. Shvahirev, O. Lopakov, V. Kosmachevskiy, and Y. Barchanova. "SIMULATION MODELING OF ADAPTIVE ROUTING UNDER EXTERNAL DESTROYING EFFECTS IN NGN NETWORKS." Odes’kyi Politechnichnyi Universytet Pratsi 3, no. 62 (2020): 101–12. http://dx.doi.org/10.15276/opu.3.62.2020.12.
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Modern subscribers of infocommunication services require a wide class of different services and applications, implying a wide variety of protocols, technologies and transmission rates. Networks are overloaded in the prevailing situation in the market of infocommunication services: they are overflowing with numerous customer interfaces, network layers and are controlled by too many control systems. High operating costs are pushing operators to look for solutions that simplify the operation, while maintaining the possibility of creating new services and ensuring the stability of existing sources of income from the provision of communication services. The NGN concept is the concept of building next-generation communication networks (Next-generation network), providing an unlimited set of services with flexible settings for their management, personalization, creation of new services through the unification of network solutions. Multiservice network is a communication network that is built in accordance with the NGN concept and provides an unlimited set of infocommunication services (VoIP, Internet, VPN, IPTV, VoD, etc.). An NGN is a packet-switched network suitable for the provision of telecommunication services and for the use of multiple broadband transport technologies with QoS enabled, in which the service-related functions are independent of the applied transport technologies. The main feature of NGN networks is the differentiation between services and transport technologies. This allows to view the network as a logically divided entity. Each layer of the network can evolve independently without affecting other layers. Inter-layer communication is based on open interfaces. The logical separation principle allows the provision of both existing and innovative services access technologies regardless of the used transport. The basic principle of the NGN concept is to separate from each other transfer and switch functions, call control functions and service control functions.
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Sanchez-Iborra, Ramon, Maria-Dolores Cano, Joel J. P. C. Rodrigues, and Joan Garcia-Haro. "An Experimental QoE Performance Study for the Efficient Transmission of High Demanding Traffic over an Ad Hoc Network Using BATMAN." Mobile Information Systems 2015 (2015): 1–14. http://dx.doi.org/10.1155/2015/217106.
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Multimedia communications are attracting great attention from the research, industry, and end-user communities. The latter are increasingly claiming for higher levels of quality and the possibility of consuming multimedia content from a plethora of devices at their disposal. Clearly, the most appealing gadgets are those that communicate wirelessly to access these services. However, current wireless technologies raise severe concerns to support extremely demanding services such as real-time multimedia transmissions. This paper evaluates from QoE and QoS perspectives the capability of the ad hoc routing protocol called BATMAN to support Voice over IP and video traffic. To this end, two test-benches were proposed, namely, a real (emulated) testbed and a simulation framework. Additionally, a series of modifications was proposed on both protocols’ parameters settings and video-stream characteristics that contributes to further improving the multimedia quality perceived by the users. The performance of the well-extended protocol OLSR is also evaluated in detail to compare it with BATMAN. From the results, a notably high correlation between real experimentation and computer simulation outcomes was observed. It was also found out that, with the proper configuration, BATMAN is able to transmit several QCIF video-streams and VoIP calls with high quality. In addition, BATMAN outperforms OLSR supporting multimedia traffic in both experimental and simulated environments.
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Xu, Yingqi, Wang-Chien Lee, Jianliang Xu, and Gail Mitchell. "Energy-Aware and Time-Critical Geo-Routing in Wireless Sensor Networks." International Journal of Distributed Sensor Networks 4, no. 4 (2008): 315–46. http://dx.doi.org/10.1080/15501320701260410.
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Volunteer forwarding, as an emerging routing idea for large scale, location-aware wireless sensor networks, has recently received significant attention. However, several critical research issues raised by volunteer forwarding, including communication collisions, communication voids, and time-critical routing, have not been well addressed by the existing work. In this paper, we propose a priority-based stateless geo-routing (PSGR) protocol that addresses these issues. Based on PSGR, sensor nodes are able to locally determine their priority to serve as the next relay node using dynamically estimated network density. This effectively suppresses potential communication collisions without prolonging routing delays. PSGR also overcomes the communication void problem using two alternative stateless schemes, rebroadcast and bypass. Meanwhile, PSGR supports routing of time-critical packets with different deadline requirements at no extra communication cost. Additionally, we analyze the energy consumption and the delivery rate of PSGR as functions of the transmission range. Finally, an extensive performance evaluation has been conducted to compare PSGR with competing protocols, including GeRaf, IGF, GPSR, flooding, and MSPEED. Simulation results show that PSGR exhibits superior performance in terms of energy consumption, routing latency, and delivery rate, and soundly outperforms all of the compared protocols.
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Alfaisaly, Noor Nateq, Suhad Qasim Naeem, and Azhar Hussein Neama. "Enhancement of WiMAX networks using OPNET modeler platform." Indonesian Journal of Electrical Engineering and Computer Science 23, no. 3 (2021): 1510. http://dx.doi.org/10.11591/ijeecs.v23.i3.pp1510-1519.
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Worldwide interoperability microwave access (WiMAX) is an 802.16 wireless standard that delivers high speed, provides a data rate of 100 Mbps and a coverage area of 50 km. Voice over internet protocol (VoIP) is flexible and offers low-cost telephony for clients over IP. However, there are still many challenges that must be addressed to provide a stable and good quality voice connection over the internet. The performance of various parameters such as multipath channel model and bandwidth over the Star trajectoryWiMAX network were evaluated under a scenario consisting of four cells. Each cell contains one mobile and one base station. Network performance metrics such as throughput and MOS were used to evaluate the best performance of VoIP codecs. Performance was analyzed via OPNET program14.5. The result use of multipath channel model (disable) was better than using the model (ITU pedestrian A). The value of the throughput at 15 dB was approximately 1600 packet/sec, and at -1 dB was its value 1300 packet/se. According to data, the Multipath channel model of the disable type the value of the MOS was better than the ITU Pedestrian A type.
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Butt, Suhail, Kamalrulnizam Bakar, Nadeem Javaid, et al. "Exploiting Layered Multi-Path Routing Protocols to Avoid Void Hole Regions for Reliable Data Delivery and Efficient Energy Management for IoT-Enabled Underwater WSNs." Sensors 19, no. 3 (2019): 510. http://dx.doi.org/10.3390/s19030510.
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The key concerns to enhance the lifetime of IoT-enabled Underwater Wireless Sensor Networks (IoT-UWSNs) are energy-efficiency and reliable data delivery under constrained resource. Traditional transmission approaches increase the communication overhead, which results in congestion and affect the reliable data delivery. Currently, many routing protocols have been proposed for UWSNs to ensure reliable data delivery and to conserve the node’s battery with minimum communication overhead (by avoiding void holes in the network). In this paper, adaptive energy-efficient routing protocols are proposed to tackle the aforementioned problems using the Shortest Path First (SPF) with least number of active nodes strategy. These novel protocols have been developed by integrating the prominent features of Forward Layered Multi-path Power Control One (FLMPC-One) routing protocol, which uses 2-hop neighbor information, Forward Layered Multi-path Power Control Two (FLMPC-Two) routing protocol, which uses 3-hop neighbor information and ’Dijkstra’ algorithm (for shortest path selection). Different Packet Sizes (PSs) with different Data Rates (DRs) are also taken into consideration to check the dynamicity of the proposed protocols. The achieved outcomes clearly validate the proposed protocols, namely: Shortest Path First using 3-hop neighbors information (SPF-Three) and Breadth First Search with Shortest Path First using 3-hop neighbors information (BFS-SPF-Three). Simulation results show the effectiveness of the proposed protocols in terms of minimum Energy Consumption (EC) and Required Packet Error Rate (RPER) with a minimum number of active nodes at the cost of affordable delay.
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Touil, Hamza, Nabil El Akkad, and Khalid Satori. "Secure and Guarantee QoS in a Video Sequence: a New Approach Based on TLS Protocol to Secure Data and RTP to Ensure Real-time Exchanges." WSEAS TRANSACTIONS ON COMMUNICATIONS 20 (April 2, 2021): 52–62. http://dx.doi.org/10.37394/23204.2021.20.7.
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The continued development of networks has significantly contributed to increasing the quantity of information available to replace old intelligence-gathering methods faster and more efficiently. For this, it is necessary to implement services that meet the consumers' requirements and measure precisely the factors that can generate obstacles to any communication, among these causes we can cite strong security and high quality of services. In this work, we implement a secure approach useful in continuous communications in a time axis (video sequence, VOIP call...), the process consists in establishing a well-secured connection between two interlocutors (the server that broadcasts the video sequence and a client) using an AES encryption key of size 256. A step of jitter check (latency variation) periodically is essential for the customer in order to make a decision: If the jitter is within the standards (compared to the tolerable value), we continue to encrypt with the AES256 key, if no, both ends must go through an automatic and uninterrupted fast renegotiation of the video to switch to a small AES key (192,128) to reduce the bandwidth on the channel, this operation must be repeated in an alternative way until the end of the communication.
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Li, Meiju, Xiujuan Du, Xin Liu, and Chong Li. "Shortest Path Routing Protocol Based on the Vertical Angle for Underwater Acoustic Networks." Journal of Sensors 2019 (July 4, 2019): 1–14. http://dx.doi.org/10.1155/2019/9145675.
Full textAbstract:
Underwater Acoustic Networks (UANs) use acoustic communication. UANs are characterized by narrow bandwidth, long delay, limited energy, high bit error rate, and dynamic network topology. Therefore, UANs call for energy-efficient and latency-minimized routing protocol. In this paper, the shortest path routing protocol based on the vertical angle (SPRVA) is proposed. In SPRVA, the forwarding node determines the best next-hop according to main priority. When the main priorities of candidate nodes are the same, the alternative priority is used. The main priority is denoted by the residual energy and angle between propagation direction and depth direction, and the alternative priority is indicated by the link quality. SPRVA selects the node along the depth direction with more residual energy and better link quality as the best next-hop. In addition, a recovery algorithm is designed to avoid nodes in void areas as forwarding nodes. Simulation results show that SPRVA improves energy efficiency and decreases end-to-end communication delay.
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Wu, Zhijun, Rong Li, Panpan Yin, and Changliang Li. "Steganalysis of Quantization Index Modulation Steganography in G.723.1 Codec." Future Internet 12, no. 1 (2020): 17. http://dx.doi.org/10.3390/fi12010017.
Full textAbstract:
Steganalysis is used for preventing the illegal use of steganography to ensure the security of network communication through detecting whether or not secret information is hidden in the carrier. This paper presents an approach to detect the quantization index modulation (QIM) of steganography in G.723.1 based on the analysis of the probability of occurrence of index values before and after steganography and studying the influence of adjacent index values in voice over internet protocol (VoIP). According to the change of index value distribution characteristics, this approach extracts the distribution probability matrix and the transition probability matrix as feature vectors, and uses principal component analysis (PCA) to reduce the dimensionality. Through a large amount of sample training, the support vector machine (SVM) is designed as a classifier to detect the QIM steganography. The speech samples with different embedding rates and different durations were tested to verify their impact on the accuracy of the steganalysis. The experimental results show that the proposed approach improves the accuracy and reliability of the steganalysis.
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Ganesh, N. "Performance Evaluation of Depth Adjustment and Void Aware Pressure Routing (DA-VAPR) Protocol for Underwater Wireless Sensor Networks." Computer Journal 63, no. 2 (2019): 193–202. http://dx.doi.org/10.1093/comjnl/bxz093.
Full textAbstract:
Abstract Underwater wireless sensor network (UWSN) has gained its popularity as a powerful technology for monitoring oceans, sea and river. The sensor node drifting along with ocean current offers 4D (space and time) monitoring for real-time underwater application. However, the main challenge arises from the underwater acoustic communication that results in high propagation delay, packet loss and overhead in the network. In order to overcome these issues, a depth adjustment and void aware pressure routing protocol is proposed for UWSN. A greedy forwarding strategy is used to forward the packet. In case a node fails to forward the packet using greedy forwarding strategy, then it immediately switches to the recovery mode. In the recovery mode, the node determines the new depth using particle swarm optimization technique. The global best value gives the new depth with minimum displacement. The void node forwards the packet with minimum displacement without any packet loss and delay.