SIP Diyalog Analitiği ve Ses Kaydı Ayrıştırılması Modeli

Year 2019, Volume: 23 Issue: 3, 831 - 839, 25.12.2019

Adnan Ozsoy Fatih Özdemir Çağdaş Kopuz

https://doi.org/10.19113/sdufenbed.578491

Abstract

VoIP ve internet teknolojisindeki gelişmeler ve sunduğu avantajlar nedeniyle, internet üzerinden iletişim çok yaygın hale gelmiştir. Mevcutta kullanılan VoIP birimlerinin özellikle ses kayıt gibi I/O ağırlıklı işlemler sırasında o anki iletişim akışında duraklamalar ve bozulmalara neden olması nedeniyle, iletişimin I/O ağırlıklı işlemler düşünülerek yeniden tasarlanması gerekliliği ortaya çıkmıştır. Bu yayında iletişimin kayıt altına alınması ve ileride kullanıcıya ayrı bir döküm olarak sunulması üzerine karşılaşılan ölçekleme yetersizliklerinin çözümü üzerine bir model sunulmaktadır. İletişimin gerçekleştiği sunucu üzerinde iletişim ile beraber yapılan ses kaydında karşılaşılan sorunlar nedeniyle ortam ağ geçidi trafiğini yansıtma yöntemi ile başka bir sunucuya alınması ve analiz edilerek ses kaydının hangi kiracıya ve kime ait olduğunun tespit edilip eşleştirilerek kayıt edilmesi, ses formatı dönüşümünün yapılması ve kullanıcı ara yüzünden dinlenebilmesi yayın kapsamında incelenmiştir. Ayrıca birden çok işleyiciye adaptif olarak ölçeklenmesine imkan veren bir mimari sunulmuştur. Sunulan modelin programlanması ve işletilmesi sonucu aynı anda işlenebilen görüşme sayısında 5,33 kata kadar bir iyileştirme elde edilmiş, kayıt işlemlerinde yatay ölçekleme sağlanabilmiştir.

Keywords

Telekomünikasyon, VoIP, Ses kaydı ayrıştırılması, SIP diyalog analitiği

Supporting Institution

Tübitak

Project Number

7180242

Thanks

Bu proje Sanayi Bakanlığı'nın Ar-Ge merkezi bünyesinde sağlanan desteklerle ve TÜBİTAK TEYDEB 7180242 numaralı 1507 Kobi-ArGe projesi kapsamında gerçekleştirilmiştir.

SIP Dialogue Analytics and Voice Record Decomposition Model

Abstract

Due to the advancements in VoIP and internet technology and the advantages they offer, communication over the internet has adopted widespread. Due to the fact that the VoIP units used in the current process cause interruptions and distortions in the communication flow during the I/O-weighted operations such as voice recording, the necessity of redesigning the communication with I/O-weighted operations has emerged. In this publication, a model is presented on the solution of scaling deficiencies encountered upon recording the communication and presenting it as a separate casting to the user in the future. Due to the problems encountered in the recording of the communication on the same server where the communication is processed as well, forwarding to another server with the method of packet mirroring is studied. Further, analysis of the voice record to identify the tenant which the recording belongs to, matching the continous call information, recording the call, and converting to appropriate audio format are shown. The model offers an architecture that allows adaptive scaling to multiple processors. As a result of the presented model, an improvement of up to 5.33 was achieved in the number of simultaneous calls that can be achieved and horizontal scaling could be achieved among call recording processes.

Keywords

Telecommunication, VoIP, Voice record decomposition, SIP dialogue analytics

Project Number

7180242

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