prikaz prve stranice dokumenta Analiza telefonskog poziva preko VoIP protokola
Access restricted to students and staff of home institution
master's thesis
Analiza telefonskog poziva preko VoIP protokola
2020. urn:nbn:hr:190:127008
Poropat, Dante
University of Rijeka
Faculty of Engineering
Department of Computer Engineering
Section of Communication Systems

Cite this document

Poropat, D. (2020). Analiza telefonskog poziva preko VoIP protokola (Master's thesis). Rijeka: University of Rijeka, Faculty of Engineering. Retrieved from https://urn.nsk.hr/urn:nbn:hr:190:127008

Poropat, Dante. "Analiza telefonskog poziva preko VoIP protokola." Master's thesis, University of Rijeka, Faculty of Engineering, 2020. https://urn.nsk.hr/urn:nbn:hr:190:127008

Poropat, Dante. "Analiza telefonskog poziva preko VoIP protokola." Master's thesis, University of Rijeka, Faculty of Engineering, 2020. https://urn.nsk.hr/urn:nbn:hr:190:127008

Poropat, D. (2020). 'Analiza telefonskog poziva preko VoIP protokola', Master's thesis, University of Rijeka, Faculty of Engineering, accessed 25 December 2024, https://urn.nsk.hr/urn:nbn:hr:190:127008

Poropat D. Analiza telefonskog poziva preko VoIP protokola [Master's thesis]. Rijeka: University of Rijeka, Faculty of Engineering; 2020 [cited 2024 December 25] Available at: https://urn.nsk.hr/urn:nbn:hr:190:127008

D. Poropat, "Analiza telefonskog poziva preko VoIP protokola", Master's thesis, University of Rijeka, Faculty of Engineering, Rijeka, 2020. Available at: https://urn.nsk.hr/urn:nbn:hr:190:127008

Please login to the repository to save this object to your list.
Metadata
TitleAnaliza telefonskog poziva preko VoIP protokola
Title (english)An Analysis of a Phone Call Quality over the VoIP Protocol
AuthorDante Poropat
MentorMiroslav Joler (mentor)
Committee memberMladen Tomić (predsjednik povjerenstva)
Committee memberGoran Mauša (član povjerenstva)
Committee memberMiroslav Joler (član povjerenstva)
GranterUniversity of Rijeka
Faculty of Engineering
(Department of Computer Engineering)
(Section of Communication Systems)
Rijeka
Defense date and country2020-09-16, Croatia
Scientific / art field,
discipline and subdiscipline		TECHNICAL SCIENCES
Computing
Abstract
U ovom radu je osmišljena VoIP infrastruktura pomoću koje su izvršena mjerenja i analiza kvalitete VoIP poziva. WebRTC je korišten za iniciranje poziva prema Asterisku koji je reproducirao glasovni zapis i tako simulirao javljanje na poziv. SBC je korišten u obliku programske podrške i služio je za autorizaciju poziva, te za mjerenje kvalitete usluge. Od VoIP protokola, za signalizaciju je korišten SIP, a za prijenos informacija RTP. Za mjernu veličinu kvalitete poziva odabrali smo MOS koji je u pravilu subjektivna ocjena koja se može izračunati i algoritamski preko R faktora. MOS je računat u svakoj sekundi poziva kako bi lakše protumačili promjenu kvalitete u odnosu na promjenu algoritamskih varijabli. Proveli smo ukupno 8 testnih scenarija s različitim vrijednostima mrežnih parametara. Najviše pažnje smo pridodavali kašnjenju paketa, gubitku paketa, odnosno broju primljenih i poslanih paketa, vremenu odziva, varijaciji u kašnjenju paketa. Pokazalo se da gubitak paketa ima najveći utjecaj na kvalitetu poziva, bilo da je on izazvan direktno ili indirektno, kašnjenjem signala i varijacijom u kašnjenju.
Abstract (english)
In this paper, we defined the VoIP infrastructure, inside which we measure and analyze call quality. WebRTC was used to initiate a call to an Asterisk, which played a voice record and thus simulated call answering. SBC was used as a software to authorize calls and to measure the quality of service. SIP was used for signaling and RTP for information transmission out of VoIP protocols. For call quality measurement, we chose MOS, which is usually a subjective rating that can be calculated algorithmically via the R factor. MOS was calculated every second of a call to more easily interpret a change in quality relative to a change in algorithmic variables. We conducted eight test scenarios with varied network parameter values. Our attention was paid to packet latency, packet loss (the ratio of sent-packages vs received-packages), round-trip time, jitter. It was found that package loss has a major impact on the call quality, whether the loss was caused directly or indirectly through the signal latency or jitter.
Keywords
Keywords (english)
Languagecroatian
URN:NBNurn:nbn:hr:190:127008
Study programmeTitle: Computer Engineering
Study programme type: university
Study level: graduate
Academic / professional title: magistar/magistra inženjer/inženjerka računarstva (magistar/magistra inženjer/inženjerka računarstva)
Type of resourceText
File originBorn digital
Access conditionsAccess restricted to students and staff of home institution
Terms of useLicence In copyright icon
Created on2021-08-26 10:12:52