Relevant bibliographies by topics / VoIP

Contents

  1. Journal articles
  2. Dissertations / Theses
  3. Books
  4. Book chapters
  5. Conference papers
  6. Reports

Academic literature on the topic 'VoIP'

Author: Grafiati
Published: 4 June 2021
Last updated: 15 June 2025

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Journal articles on the topic "VoIP"

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Lundblad, Henrik, Gerald Q. Maguire, Charlotte Karlsson-Thur, et al. "Using PET/CT Bone Scan Dynamic Data to Evaluate Tibia Remodeling When a Taylor Spatial Frame Is Used: Short and Longer Term Differences." BioMed Research International 2015 (2015): 1–11. http://dx.doi.org/10.1155/2015/574705.

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Eighteen consecutive patients, treated with a Taylor Spatial Frame for complex tibia conditions, gave their informed consent to undergo Na18F−PET/CT bone scans. We present a Patlak-like analysis utilizing an approximated blood time-activity curve eliminating the need for blood aliquots. Additionally, standardized uptake values (SUV) derived from dynamic acquisitions were compared to this Patlak-like approach. Spherical volumes of interest (VOIs) were drawn to include broken bone, other (normal) bone, and muscle. TheSUVm(t)(m=max, mean) and a series of slopes were computed as(SUVm(ti)-SUVm(tj))/(ti-tj), for pairs of time valuestiandtj. A Patlak-like analysis was performed for the same time values by computing((VOIp(ti)/VOIe(ti))-(VOIp(tj)/VOIe(tj)))/(ti-tj), wherep= broken bone, other bone, and muscle ande= expected activity in a VOI. Paired comparisons between Patlak-like andSUVmslopes showed good agreement by both linear regression and correlation coefficient analysis (r=84%,rs=78%-SUVmax,r=92%, andrs=91%-SUVmean), suggesting static scans could substitute for dynamic studies. Patlak-like slope differences of 0.1 min−1or greater between examinations andSUVmaxdifferences of ~5 usually indicated good remodeling progress, while negative Patlak-like slope differences of −0.06 min−1usually indicated poor remodeling progress in this cohort.
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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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ANDRIANTO, HERI, DANIEL SETIADIKARUNIA, and HENDRY RAHARJO. "Evaluasi Kinerja GSM VoIP Gateway pada Sistem IP PBX." ELKOMIKA: Jurnal Teknik Energi Elektrik, Teknik Telekomunikasi, & Teknik Elektronika 9, no. 3 (2021): 731. http://dx.doi.org/10.26760/elkomika.v9i3.731.

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ABSTRAKGSM VoIP Gateway digunakan untuk menghubungkan jaringan VoIP dengan jaringan GSM sehingga memungkinkan VoIP client melakukan komunikasi dengan VoIP client lain melalui jaringan GSM sehingga biaya komunikasi dapat ditekan. Pada penelitian ini, telah dirancang dan direalisasikan sistem IP PBX yang dihubungkan ke jaringan GSM menggunakan GSM VoIP Gateway. Evaluasi kinerja GSM VoIP Gateway pada sistem IP PBX dilakukan dengan mengamati nilai parameter Quality of Service (QoS). Komunikasi antara VoIP client dengan GSM VoIP Gateway dikategorikan pada kualitas layanan VoIP yang baik karena memiliki nilai rata-rata jitter ≤ 5,7 ms, packet loss ≤ 0,18% dan delay ≤ 9,41 ms. Komunikasi antara softphone SIPdroid dengan GSM VoIP Gateway memiliki nilai rata-rata jitter 22,58 ms, paket loss 48,68%, dan delay 14,54 ms, hal ini disebabkan karena komunikasi VoIP menggunakan koneksi WiFi. Selain itu perbedaan spesifikasi perangkat keras dan perangkat lunak juga turut mempengaruhi nilai parameter QoS.Kata kunci: GSM VoIP Gateway, IP PBX, VoIP ABSTRACTGSM VoIP Gateway is used to connect the VoIP network to the GSM network, allowing VoIP clients to communicate with other VoIP clients via the GSM network therefore the communication costs can be reduced. In this research, an IP PBX system connected to a GSM network using a GSM VoIP Gateway has been designed and realized. Performance evaluation of the GSM VoIP Gateway on the IP PBX system is carried out by observing the value of the Quality of Service (QoS) parameter. Communication between the VoIP client and GSM VoIP Gateway is categorized as a good quality VoIP service because it has an average value of jitter ≤ 5.7 ms, packet loss ≤ 0.18% and delay ≤ 9.41 ms. Communication between the SIPdroid softphone and the GSM VoIP Gateway has an average jitter value of 22.58 ms, a packet loss of 48.68%, and a delay of 14.54 ms, due to VoIP communication uses a WiFi connection. In addition, differences on hardware and software specifications also affect the value of QoS parameters.Keywords: GSM VoIP Gateway, IP PBX, VoIP
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Child, M. "Briefing: VoIP." ITNOW 47, no. 1 (2004): 28. http://dx.doi.org/10.1093/combul/bwi011.

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Johnston, Elizabeth. "Editorial [VoIP]." IEEE Potentials 26, no. 1 (2007): 3. http://dx.doi.org/10.1109/mp.2007.343012.

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Gold, Steve. "Securing VoIP." Network Security 2012, no. 3 (2012): 14–17. http://dx.doi.org/10.1016/s1353-4858(12)70046-6.

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Soupionis, Yannis, and Dimitris Gritzalis. "Hacking VoIP." Computers & Security 32 (February 2013): 267. http://dx.doi.org/10.1016/j.cose.2012.09.006.

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Ebbinghaus, R. "VoIP lessons." Communications Engineer 1, no. 5 (2003): 28–31. http://dx.doi.org/10.1049/ce:20030505.

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Materna, B. "VoIP insecurity." Communications Engineer 4, no. 5 (2006): 39–42. http://dx.doi.org/10.1049/ce:20060507.

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Bhebhe, Leo, and Rauli Parkkali. "VoIP Performance over HSPA with Different VoIP Clients." Wireless Personal Communications 58, no. 3 (2010): 613–26. http://dx.doi.org/10.1007/s11277-010-0125-2.

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Dissertations / Theses on the topic "VoIP"

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Schildt, Holger. "VoIP mit IAX." Universitätsbibliothek Chemnitz, 2004. http://nbn-resolving.de/urn:nbn:de:swb:ch1-200400518.

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Workshop "Netz- und Service-Infrastrukturen" Das Inter-Asterisk eXchange (IAX)-Protokoll ermöglicht eine unproblematische Kommunikation zwischen IAX-fähigen VoIP-Systemen. In der Präsentation zu dem Vortrag werden das Protokoll vorgestellt und die Vorteile von IAX skizziert.
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Correia, Ricardo João Luís Marques. "Scrambler para VoIP." Master's thesis, Universidade de Aveiro, 2009. http://hdl.handle.net/10773/7364.

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Mestrado em Engenharia Electrónica e Telecomunicações<br>A necessidade de se preservar a confidencialidade numa conversa telefónica é um facto dos nossos dias. Pretende-se com este trabalho dar uma resposta a este problema propondo um sistema original de scrambling do sinal no domínio das frequências. O sistema inclui uma troca de chaves públicas que geram uma chave secreta comum entre o emissor e o receptor, baseado no método de Diffie-Hellman. Além da implementação apresentam-se resultados de testes efectuados sobre o sistema de scrambling proposto associado a vários codecs de uso comum.<br>The need to preserve confidentiality in a telephone conversation is, nowadays, a motive of concern for most of us. This work pretends to address this issue by proposing an original signal scrambling system in the frequency domain. The system includes a public key exchange which generates a common secret key, based on the Diffie-Hellman method, at the transmitter and the receiver. The implementation and some results of the proposed solution using common conventional codecs are presented.
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Pini, Fabio. "VoIP su BlackBerry." Bachelor's thesis, Alma Mater Studiorum - Università di Bologna, 2010. http://amslaurea.unibo.it/1631/.

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Lembard, Tomáš. "Speciální aplikace VoIP." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2011. http://www.nusl.cz/ntk/nusl-219188.

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The aim of this master's thesis is suggestion and following realization of voice transmission over the local network equipment and a description of used circuits and solutions in terms of hardware and software. This thesis deals with digitization of low-frequency signals, the structure of IP and UDP protocols, implementation of TCP/IP stack cIPS
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Kristensveen, David. "Sikkerhet i VoIP-portnere." Thesis, Norwegian University of Science and Technology, Department of Telematics, 2006. http://urn.kb.se/resolve?urn=urn:nbn:no:ntnu:diva-10267.

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<p>Session Initiation Protocol(SIP) er i ferd med å bli den ledende signaleringsprotokollen i forbindelse med IP-telefoni. SIP benyttes til å initiere, modifisere og terminere interaktive sesjoner. Arkitekturens to hovedkomponenter er servere og brukeragenter. De ulike brukeragentene utveksler forespørsler og tilhørende responsmeldinger. Etter at en sesjon er satt opp av SIP benyttes Real Time Protocol(RTP) til å overføre data i forbindelse med selve samtalen. RTP benytter dynamisk valgte portnumre. Disse utveksles på forhånd mellom brukeragentene ved hjelp av Session Description Protocol(SDP) i meldingskroppen til SIP-meldingene. Når brannmurer eller Network Address Translation(NAT) benyttes sammen med IP-telefoni er det et som regel et problem at IP-adresser og portnumrene som skal benyttes av brukeragentene omskrives. Adresseinformasjonen som er utvekslet på forhånd vil derfor bli ugyldig. Det finnes flere ulike metoder for å tilnærme seg disse problemene. To metoder fra IETF er Simpel Traversal of UDP through NAT(STUN) og Traversal using Relay NAT(TURN). Session Border Controllers(SBC) er lukkede kommersielle nettverksløsninger som benyttes til å løse mange av de samme problemene. Ulempen med SBC er at dette er kostbare løsninger og at prinsippet om at SIP skal være en åpen protokoll brytes. Innefor IP-telefoni ser en utvikling der tilbydere eller organisasjoner har infrastrukturen på plass for å realisere en IP-telefonitjeneste på vegne av sine egne brukere, mens samtrafikk med andre er nødt til å foregå ved hjelp av PSTN-nettet. For å realisere en slik samtrafikk er en nødt til å benytte en Gateway. En slik Gateway har oftest en todelt funksjonalitet. Først må det oversettes mellom signaleringen i IP-telefoni(SIP) og signaleringen i PSTN(som regel ISUP). Dette utføres av en signaliseringsgsgateway(SG). Deretter må mediastrømmen oversettes av en mediagateway(MG) fra RTP til det aktuelle formatet som benyttes av det linjesvitsjede nettverket. En Media Gateway Controller(MGC) benyttes for å samkjøre MG og SG. Telephony Routing over IP(TRIP) er en protokoll som kan benyttes av tjenestetilbydere eller organisasjoner for å utveksle rutingtabeller for sine respektive gatewayer. En annen mekanisme for PSTN til IP samtrafikk er SIP for Telephones(SIP-T). Her kan PSTN-signaleringen enten oversettes til SIP eller pakkes inn i SIP-meldinger. En av hovedgrunnene til at PSTN benyttes til ruting i IP-telefon skyldes at det ikke har eksistert noen fullgod erstatter for Signaleringssystem nummer syv(SS7) i forbindelse med IP-telefoni. E.164 Number Mapping(ENUM) er en metode for å oversette E.164-numre til domenenavn ved hjelp av DNS. Ved en slik løsning vil en være i stand til å realisere samtrafikk mellom ulike typer IP-telefoninettverk uten å benytte PSTN. I motsetning til tradisjonelle løsninger der mange i dag benytter to ulike nettverk for data- og taletjenester vil en fremover kunne se en utvikling der alle tjenester er basert på IP og PSTN vil bli overflødig. Samtidig ser en utvikling der Internett løsninger som Skype blir stadig mer populære. Det er også muligheter for å benytte SIP til tilsvarende løsninger, men her er det fortsatt en del uenighet i om hvordan slike løsninger konkret skal realiseres. The IP Multimedia Subsystem(IMS) er et rammeverk som skal sørge for ytterligere konvergens mellom telefoni- og datatjenester. Telefonikunder skal tilbys nye multimediatjenester på apllikasjonsnivå. Kjernenettverket i IMS skal benytte SIP-baserte løsninger. IMS omtales ofte som neste generasjons nettverk. Etter hvert som 802.11 nettverk er blitt mer utbredt er det et voksende marked for IP-telefoni over 802.11. Unlicensed Mobile Access(UMA) er en teknologi som lar brukeren benytte mobiltelefonen over 802.11 eller Bluetooth. Samtidig tillates det handover mot det mobile nettverket når dette er nødvendig. Sikkerhet i trådløse nettverk er et område under stadig utvikling. Ved innføring av 802.11i vil en få sterkere mekanismer for sikkerhet i form av Extended Authentication Protocol(EAP) for autentisering og Advanced Encryption Standard(AES) for kryptering. Ved introduksjonen av ENUM oppstår det nye sikkerhetsutfordringer. Et system med billigere telefoni og enkel tilgangs til brukerlokasjoner vil potensielt kunne føre til mer Spam over IP-telefoni(SPIT). Ellers er DNS systemet utsatt for mange potensielle trusler.</p>
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Saad, Amna. "Secure VoIP performance measurement." Thesis, Loughborough University, 2013. https://dspace.lboro.ac.uk/2134/13426.

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This project presents a mechanism for instrumentation of secure VoIP calls. The experiments were run under different network conditions and security systems. VoIP services such as Google Talk, Express Talk and Skype were under test. The project allowed analysis of the voice quality of the VoIP services based on the Mean Opinion Score (MOS) values generated by Perceptual valuation of Speech Quality (PESQ). The quality of the audio streams produced were subjected to end-to-end delay, jitter, packet loss and extra processing in the networking hardware and end devices due to Internetworking Layer security or Transport Layer security implementations. The MOS values were mapped to Perceptual Evaluation of Speech Quality for wideband (PESQ-WB) scores. From these PESQ-WB scores, the graphs of the mean of 10 runs and box and whisker plots for each parameter were drawn. Analysis on the graphs was performed in order to deduce the quality of each VoIP service. The E-model was used to predict the network readiness and Common vulnerability Scoring System (CVSS) was used to predict the network vulnerabilities. The project also provided the mechanism to measure the throughput for each test case. The overall performance of each VoIP service was determined by PESQ-WB scores, CVSS scores and the throughput. The experiment demonstrated the relationship among VoIP performance, VoIP security and VoIP service type. The experiment also suggested that, when compared to an unsecure IPIP tunnel, Internetworking Layer security like IPSec ESP or Transport Layer security like OpenVPN TLS would improve a VoIP security by reducing the vulnerabilities of the media part of the VoIP signal. Morever, adding a security layer has little impact on the VoIP voice quality.
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Sitolino, Claudio Luis. "VOIP : um estudo experimental." reponame:Biblioteca Digital de Teses e Dissertações da UFRGS, 2001. http://hdl.handle.net/10183/3182.

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Voz sobre IP (VoIP) é uma tecnologia que permite a digitalização e a codificação da voz e o empacotamento em pacotes de dados IP para a transmissão em uma rede que utilize o protocolo TCP/IP. Devido ao volume de dados gerados por uma aplicação VoIP, esta tecnologia se encontra em funcionamento, em redes corporativas privadas. Mas se a rede base para o transporte desta aplicação for a Internet, certamente, não deve ser utilizada para fins profissionais, pois o TCP/IP não oferece padrões de QoS (Qualidade de Serviço) comprometendo desta forma a qualidade da voz. A qualidade da voz fica dependente do tráfego de dados existentes no momento da conversa. Para realizar um projeto de VoIP é necessário conhecer todo o tráfego existente na rede e verificar o quanto isto representa em relação à banda total da rede. Também se deve conhecer o tipo de aplicação que se deseja implantar, verificando a banda a ser utilizada por esta, e então projetar como a rede deverá ser estruturada. Para auxiliar no projeto de VoIP, pretende-se mostrar o que está sendo desenvolvido para que o protocolo TCP/IP ofereça QoS e uma ferramenta para análise do tráfego de voz sobre redes TCP/IP e também análises dos resultados obtidos em experimentos simulando diversas situações práticas.
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Бубнов, Ігор Васильович, Игорь Васильевич Бубнов, Ihor Vasylovych Bubnov та Н. В. Москаленко. "Практическое использование технологии VoIP". Thesis, Издательство СумГУ, 2008. http://essuir.sumdu.edu.ua/handle/123456789/7477.

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Нами для проведения работ по использованию технологии VoIP была избрана бесплатно распространяемая программа Echolink используемая в радиолюбительской практике. Разработчиком системы является Джонатан Тейлор (K1RFD). При цитировании документа, используйте ссылку http://essuir.sumdu.edu.ua/handle/123456789/7477
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Kulička, Vojtěch. "VoIP in Jabber Client." Master's thesis, Vysoké učení technické v Brně. Fakulta informačních technologií, 2011. http://www.nusl.cz/ntk/nusl-237034.

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Práce se zabývá možnostmi implementace VoIP do existujícího XMPP programu se sdílenou tabulí. Analyzuje možnosti využití současných technologií pro podporu VoIP.  Cílem je nahrazení stávajících komunikačních knihoven klienta za telepathy. Dále také přidání VoIP.
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Havelka, Ondřej. "Analyzátor kvality VoIP hovorů." Master's thesis, Vysoké učení technické v Brně. Fakulta informačních technologií, 2011. http://www.nusl.cz/ntk/nusl-237055.

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This master thesis deals with the design and implementation of an application for analyzing Voice over IP quality using NetFlow. In the beginning, there is summarized basic information about VoIP technology and NetFlow - its principles, the most used protocols, factors that have influence on call quality and call quality rating methods. Later there is presented proposal of application and then described its implementation. The created application was tested on samples, which simulate calls in network with delays and packet-loss. Within testing was made the comparison with commercial application and the results are discussed.
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Books on the topic "VoIP"

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Fischer, Jörg. VoIP-Praxisleitfaden. Carl Hanser Verlag GmbH & Co. KG, 2008. http://dx.doi.org/10.3139/9783446415973.

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W, Rittinghouse John, ed. VoIP security. Elsevier, 2005.

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Hersent, Olivier, Jean-Pierre Petit, and David Gurle. Beyond VoIP Protocols. John Wiley & Sons, Ltd, 2005. http://dx.doi.org/10.1002/0470023643.

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Wolf, Karl Heinz, and Richard Barnes. VoIP Emergency Calling. John Wiley & Sons, Ltd, 2010. http://dx.doi.org/10.1002/9780470976975.

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Kelly, Timothy V. VoIP For Dummies. John Wiley & Sons, Ltd., 2005.

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Petit, Jean-Pierre. Beyond VoIP Protocols. John Wiley & Sons, Ltd., 2005.

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B.C.) IEEE Workshop on VoIP Management and Security (1st 2006 Vancouver. The 1st IEEE Workshop on VoIP Management and Security: VoIP MaSe 2006 : securing and managing VoIP communications. Edited by Niccolini Saverio, State Radu 1972-, Schulzrinne Henning, IEEE Communications Society, and International Federation for Information Processing. IEEE, 2006.

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Raake, Alexander. Speech Quality of VoIP. John Wiley & Sons, Ltd, 2006. http://dx.doi.org/10.1002/9780470033005.

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Flanagan, William A. VoIP and Unified Communications. John Wiley & Sons, Inc., 2012. http://dx.doi.org/10.1002/9781118166048.

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Technologies, Inc Javvin. VOIP technology quick guide. Javvin, 2008.

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Book chapters on the topic "VoIP"

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Wu, Chiung-Yi, Kuo-Ping Wu, Jason Shih, and Hahn-Ming Lee. "VoIPS: VoIP Secure Encryption VoIP Solution." In Communications in Computer and Information Science. Springer Berlin Heidelberg, 2011. http://dx.doi.org/10.1007/978-3-642-23948-9_11.

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Dantos, Christopher, and John Mason. "Securing Voip." In Computer Security Handbook. John Wiley & Sons, Inc., 2015. http://dx.doi.org/10.1002/9781118851678.ch34.

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Badach, Anatol. "VoIP-Sicherheit." In Voice over IP – Die Technik, 5th ed. Carl Hanser Verlag GmbH & Co. KG, 2022. http://dx.doi.org/10.3139/9783446471504.011.

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Fischer, Jörg, and Christian Sailer. "VoIP-Analyse." In VoIP Praxisleitfaden. Carl Hanser Verlag GmbH & Co. KG, 2016. http://dx.doi.org/10.3139/9783446448148.016.

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Fischer, Jörg, and Christian Sailer. "Einleitung." In VoIP Praxisleitfaden. Carl Hanser Verlag GmbH & Co. KG, 2016. http://dx.doi.org/10.3139/9783446448148.001.

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Fischer, Jörg, and Christian Sailer. "Infrastrukturen im VoIP-Umfeld." In VoIP Praxisleitfaden. Carl Hanser Verlag GmbH & Co. KG, 2016. http://dx.doi.org/10.3139/9783446448148.002.

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Fischer, Jörg, and Christian Sailer. "Netze, QoS, Pakete und Bandbreite." In VoIP Praxisleitfaden. Carl Hanser Verlag GmbH & Co. KG, 2016. http://dx.doi.org/10.3139/9783446448148.003.

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Fischer, Jörg, and Christian Sailer. "Nummern, Adressen und Namen." In VoIP Praxisleitfaden. Carl Hanser Verlag GmbH & Co. KG, 2016. http://dx.doi.org/10.3139/9783446448148.004.

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Fischer, Jörg, and Christian Sailer. "Protokolle und Dienste für VoIP." In VoIP Praxisleitfaden. Carl Hanser Verlag GmbH & Co. KG, 2016. http://dx.doi.org/10.3139/9783446448148.005.

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Fischer, Jörg, and Christian Sailer. "Leistungsmerkmale." In VoIP Praxisleitfaden. Carl Hanser Verlag GmbH & Co. KG, 2016. http://dx.doi.org/10.3139/9783446448148.006.

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Conference papers on the topic "VoIP"

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Charran, G. N., Kayilaivendhan S, Sivabalan, T. Rajalakshmi, and S. Vasanthadev Suryakala. "VOIP Access Vision." In 2025 International Conference on Computing and Communication Technologies (ICCCT). IEEE, 2025. https://doi.org/10.1109/iccct63501.2025.11018918.

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Cruz, Bruno, and Bruno Sousa. "RiVS: Reputation in VoIP Systems." In 11th International Conference on Information Systems Security and Privacy. SCITEPRESS - Science and Technology Publications, 2025. https://doi.org/10.5220/0013133300003899.

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Radman, Pedram, Jaipal Singh, Marc Domingo, Joan Arnedo, and Alex Talevski. "VoIP." In the 8th International Conference. ACM Press, 2010. http://dx.doi.org/10.1145/1971519.1971532.

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"The 1st IEEE Workshop on VoIP Management and Security-VoIP MaSe'06." In The 1st IEEE Workshop on VoIP Management and Security-VoIP MaSe'06. IEEE, 2006. http://dx.doi.org/10.1109/voipms.2006.1638124.

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Oliveira, Carlos Henrique Rodrigues de, Luís Carlos Costa Fonseca, Caio De Castro Torres, et al. "VoIP University Solution: VoIP UEMA Project." In Anais Estendidos do Simpósio Brasileiro de Redes de Computadores e Sistemas Distribuídos. Sociedade Brasileira de Computação - SBC, 2024. http://dx.doi.org/10.5753/sbrc_estendido.2024.1645.

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VoIP University is a solution to enable the development of communication projects based on IP protocol. This paper presents a low-cost deployment called VoIP UEMA, the first project of VoIP University solution. The technical feasibility of deploying this project was checked in a proof of concept. The concept was proven showing it is possible to create user extensions with information such as name, ID, job role, course, center, and campus get information of all registered users in the academic and administration system server and make voice and video calls through applications for Windows, Android, and iOS. VoIP UEMA project innovates because there is no similar solution in the literature to quadruple play communication that allows integration with any corporate system database to create and manage the user extension list. The low-cost aspect was justified by a financial analysis showing the annual phone bill expense dropped, representing savings of more than 97%.
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Sengar, H., R. Dantu, and D. Wijesekera. "Securing VoIP and PSTN from integrated signaling network vulnerabilities." In The 1st IEEE Workshop on VoIP Management and Security-VoIP MaSe'06. IEEE, 2006. http://dx.doi.org/10.1109/voipms.2006.1638116.

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Kong, L., V. B. Balasubramaniyan, and M. Ahamad. "A lightweight scheme for securely and reliably locating SIP users." In The 1st IEEE Workshop on VoIP Management and Security-VoIP MaSe'06. IEEE, 2006. http://dx.doi.org/10.1109/voipms.2006.1638117.

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Rippon, W. J. "Threat assessment of IP based voice systems." In The 1st IEEE Workshop on VoIP Management and Security-VoIP MaSe'06. IEEE, 2006. http://dx.doi.org/10.1109/voipms.2006.1638118.

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Abdelnur, H., V. Cridlig, R. State, and O. Festor. "VoIP security assessment: methods and tools." In The 1st IEEE Workshop on VoIP Management and Security-VoIP MaSe'06. IEEE, 2006. http://dx.doi.org/10.1109/voipms.2006.1638119.

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Okabe, T., T. Kitamura, and T. Shizuno. "Statistical traffic identification method based on flow-level behavior for fair VoIP service." In The 1st IEEE Workshop on VoIP Management and Security-VoIP MaSe'06. IEEE, 2006. http://dx.doi.org/10.1109/voipms.2006.1638120.

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Reports on the topic "VoIP"

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Mentsiev, A., and Kh Supaeva. VoIP techniques. Ljournal, 2019. http://dx.doi.org/10.18411/2019-2019-2019-00001.

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เบญจพลกุล, วาทิต. การพัฒนาระบบโทรศัพท์ผ่านอินเทอร์เน็ต (VoIP) บนพื้นฐานของโพรโทคอลเริ่มต้นเซสชัน (ปีที่2). จุฬาลงกรณ์มหาวิทยาลัย, 2010. https://doi.org/10.58837/chula.res.2010.37.

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ปัจจุบันระบบโทรศัพท์ผ่านอินเทอร์เน็ตมีแนวโน้มเข้ามาแทนที่ระบบโทรศัพท์แบบ PSTN เพิ่มมากขึ้น ทั้งนี้เนื่องจากข้อได้เปรียบทางด้านค่าใช้จ่าย ประสิทธิภาพในการใช้ช่องสัญญาณ ความสะดวกในการใช้งาน และการบำรุงรักษา อย่างไรก็ตาม เครื่องโทรศัพท์ผ่านอินเทอร์เน็ตดังกล่าวต้องนำเข้าจากต่างประเทศ โครงการวิจัยนี้จึงได้พัฒนาเครื่องโทรศัพท์ผ่านอินเทอร์เน็ตต้นแบบ เพื่อเป็นแนวทางในการผลิตเครื่องโทรศัพท์ผ่านอินเทอร์เน็ตต้นทุนต่ำขึ้นใช้เองภายในประเทศ โดยในการพัฒนาส่วนของฮาร์ดแวร์ได้ใช้ไมโครคอนโทรลเลอร์ตระกูล 8051 รุ่น AT89C51ED2 ชิพควบคุมอีเทอร์เน็ต RTL8019as และชิพเข้ารหัสเสียง MC145480 เป็นส่วนประกอบหลักในส่วนของซอฟต์แวร์ได้พัฒนาส่วนติดต่อกับผู้ใช้ ส่วนติดต่อกับชิพอีเทอร์เน็ต ส่วนควบคุมข้อมูลเสียงและส่วนรองรับโพรโทคอล ICMP, ARP, IP, UDP, RTP, และ SIP ซึ่งเป็นโพรโทคอลหลักสำหรับเครื่องโทรศัพท์ผ่านอินเทอร์เน็ตพื้นฐาน เครื่องโทรศัพท์ผ่านอินเทอร์เน็ตต้นแบบที่ได้พัฒนาขึ้นได้รับการตรวจสอบความถูกต้องของการทำงานโดยการทดสอบความถูกต้องของสัญญาณ Signaling ต่างๆที่เกิดขึ้นระหว่างการทำงานของเครื่องโทรศัพท์โดยอาศัยโพรโทคอลเริ่มต้นเซสชัน และจากผลการทดสอบพบว่า เครื่องโทรศัพท์ที่ได้รับการพัฒนาขึ้นนี้สามารถทำงานฟังก์ชันพื้นฐานได้แก่ การสร้างเซสชัน การยกเลิกเซสชัน การปฏิเสธเซสชันและการสิ้นสุดเซสชันได้อย่างถูกต้องตามมาตรฐานของโพรโทคอลเริ่มต้นเซสชัน ทั้งการต่อถึงกันโดยตรงและการต่อใช้งานและการต่อใช้งานผ่านเครื่องแม่ข่าย
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Uzelac, A., and Y. Lee, eds. Voice over IP (VoIP) SIP Peering Use Cases. RFC Editor, 2011. http://dx.doi.org/10.17487/rfc6405.

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Thompson, Gardner W. An Entropy-based Approach to Detecting Anomalies in Voice over Internet Protocol (VoIP) Traffic. Defense Technical Information Center, 2010. http://dx.doi.org/10.21236/ada532059.

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พิมพ์พิณ, อลงกรณ์, та ณัฐเดช เฟื่องวรวงศ์. การพัฒนาเทคนิคการวัดสำหรับการไหลสองเฟสระหว่างของเหลวและก๊าซ. จุฬาลงกรณ์มหาวิทยาลัย, 2010. https://doi.org/10.58837/chula.res.2010.42.

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การไหลแบบสองเฟสเป็นการไหลที่สามารถพบได้ทั่วไปในอุตสาหกรรมหลากหลายประเภท ซึ่งนักวิจัยและวิศวกรออกแบบต้องเข้าใจถึงความรู้พื้นฐานของการไหลสองเฟสอย่างลึกซึ้งเพื่อการออกแบบ การควบคุม รวมถึงการปรับปรุงสมรรถนะในระบบต่าง ๆ เหล่านั้นเพื่อทำให้มีประสิทธิภาพสูงขึ้น ในงานวิจัยนี้จึงได้พยายามพัฒนาเทคนิคการวัด 2 แบบคือ เทคนิค Wire Mesh Tomography (WMT) และเทคนิคเลเซอร์ไดโอด สำหรับเทคนิค WMT ใช้หลักการของการวัดความแตกต่างการนำไฟฟ้าของของไหลทั้งหน้าตัดการไหล ในงานวิจัยนี้ เราได้พัฒนาโปรแกรมคอมพิวเตอร์ เพื่อคำนวณพารามิเตอร์ที่สำคัญจากข้อมูลที่บันทึกได้จาก WMT เพื่อเปรียบเทียบกับข้อมูลที่ได้จากการถ่ายภาพ โดยพารามิเตอร์เหล่านี้ประกอบด้วย Local void fraction ความเร็วของฟองก๊าซและขนาดฟองก๊าซ เมื่อเปรียบเทียบกันแล้ว พบว่าข้อมูล Void fraction เฉลี่ยทั้งปริมาตรที่สนใจในช่วง Void fraction ไม่เกิน 9% มีค่าความคลาดเคลื่อนอยู่ในช่วงระหว่าง ±20% ความเร็วฟองก๊าซเฉลี่ยทั้งปริมาตรที่สนใจในช่วงระหว่าง 250-350% mm/s มีค่าความคลาดเคลื่อนอยู่ในช่วงระหว่าง ±10% และขนาดฟองก๊าซเฉลี่ยทั้งปริมาตรที่สนใจในช่วงขนาดฟองก๊าซระหว่าง 2-8 mm มีค่าความคลาดเคลื่อนอยู่ในช่วงระหว่าง ±20 %สำหรับเทคนิคเลเซอร์ไดโอดอาศัยหลักการที่แสงจะหักเหไปเมื่อผ่านรอยต่อระหว่างตัวกลางคนละชนิดที่มีดัชนีหักเหทางแสงไม่เท่ากัน และความต่างศักย์จากวงจรโฟโต้ไดโอดจะแปรผัน ตามปริมาณพลังงานของเลเซอร์ที่มาตกกระทบไดโอด ซึ่งเราได้ทำการสอบเทียบอุปกรณ์กับรัศมีความโค้งของฟองอากาศขนาดต่างๆ โดยการสร้างแบบจำลองฟองอากาศขึ้นมาจาก Polydimethylsiloxane (PDMS) ซึ่งเป็นแนวทางใหม่ของการสอบเทียบแบบหนึ่ง จากผลการสอบเทียบเราทราบระยะจากปลายยอดฟองที่ความต่างศักย์ไฟฟ้าลดลงจนใกล้ศูนย์ โดยระยะดังกล่าวจะแปรผกผันกับขนาดของรัศมีความโค้งของผิวฟองอากาศซึ่งสามารถสร้างเป็นสมการที่ใช้อธิบายความสัมพันธ์ระหว่างระยะดังกล่าวและรัศมีความโค้งของผิวฟองอากาศได้ หลังจากนั้นหากนำเอาความสัมพันธ์ระหว่างค่าความต่างศักย์ไฟฟ้าที่ระยะจากยอดฟองต่าง ๆ มาทำ normalization ด้วยความต่างศักย์เมื่อลำเลเซอร์อยู่ที่ยอดฟองสำหรับค่าความต่างศักย์ไฟฟ้า และด้วยระยะทั้งหมดที่ความต่างศักย์ลดลงจนเป็นศูนย์สำหรับระยะทางแล้ว ความสัมพันธ์ในลักษณะนี้ของทุกขนาดฟองอากาศจะสอดคล้องกัน ดังนั้นหากนำเอาอุปกรณ์ไปวัดในการไหลจริง ๆ สำหรับกรณีที่ฟองอากาศเคลื่อนที่ด้วยความเร็วคงที่ ลักษณะการลดลงของสัญญาณของทุกขนาดฟองอากาศตามเวลา (กรณีความเร็วคงที่ ระยะทางและเวลาจะสัมพันธ์กันโดยตรง) ก็จะสอดคล้องกันหมด อย่างไรก็ตามในงานวิจัยนี้ได้ลองนำเอาอุปกรณ์ไปวัดการไหลของฟองอากาศในของเหลวจริงด้วย แต่ผลที่ได้มีความผิดพลาดมากซึ่งอาจจะเกิดจากการควบคุมการทดลองที่ยังทำได้ไม่ดีรวมทั้งความเร่งของการเคลื่อนที่ของฟองอากาศที่อาจจะมีค่าค่อนข้างสูง
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Cook, B. The Void Specification. Office of Scientific and Technical Information (OSTI), 2005. http://dx.doi.org/10.2172/878188.

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Kong, Zhihao, Aritro Roy Mitra, and Luna Lu. Developing AI-Assisted In-Situ NDT Method for Air-Void Distribution Testing in Fresh and Hardened Concrete. Purdue University, 2024. http://dx.doi.org/10.5703/1288284317745.

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Understanding the air void content in concrete is crucial since it significantly influences the durability and strength of the material, especially in environments susceptible to freeze-thaw cycles. This report introduces an advanced nondestructive testing (NDT) method for the in-situ detection of air voids in concrete by employing diffusive ultrasound. Focusing on the ultrasound attenuation coefficient, this research established a strong correlation with key air void metrics, including the volumetric ratio and spacing factor, as outlined in ASTM C457. The study also undertook a comparative analysis of ASTM C457 methods B and C, revealing the instrument-dependent variability in measuring air voids. One pivotal discovery was that ultrasound attenuation in concrete is majorly influenced by air voids and aggregates, with a relatively minor contribution from cement. This methodology not only offers a novel approach for accurately assessing air void content but also enables visualization of air void distribution in concrete infrastructures like pavements. The findings of this research offer insights for enhancing concrete quality control and ensuring structural integrity in construction, particularly when the in-place air voids conditions are of interest.
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Woods, D. T., H. Robey, and P. Stry. Analysis of Shock-Void Experiment. Office of Scientific and Technical Information (OSTI), 2003. http://dx.doi.org/10.2172/15003913.

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Salko, Robert K., Chris Gosdin, Maria N. Avramova, and Marcus Gergar. CTF Void Drift Validation Study. Office of Scientific and Technical Information (OSTI), 2015. http://dx.doi.org/10.2172/1342656.

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Reding, Derek J., Pavol Stofko, Robert J. Dorgan, and Michael E. Nixon. Void Growth and Coalescence Simulations. Defense Technical Information Center, 2013. http://dx.doi.org/10.21236/ada593137.

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