Performance of Various Codecs Related to Jitter Buffer Variation in VoIP Using SIP

Iwan Handoyo Putro
Journal article Jurnal Teknik Elektro Universitas Kristen Petra • September 2008

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(English, 6 pages)


Briefly speaking, there are two popular Voice over Internet Protocol (VoIP) standards, H.323 and Session Initiation Protocol (SIP). The first standard was designed by ITU and has become the basis for the widespread implementation of VoIP systems although it was not specifically designed for it. The second standard, SIP, was proposed by IETF and it is designed to connect, communicate and exchange data with the internet applications. In order to deliver voice conversation through packet-switching networks, codecs (coder-decoder) should be implemented to compress and later decompress those packets. In this paper, some of compression algorithms will be compared and analyzed based on its performances in SIP based VoIP system. The codecs that was used in this experiment are SJ Lab GSM 6.10, SJ Lab iLBC-30ms, SJ Lab iLBC-20ms, Microsoft CCITT G.711 A-law and Microsoft CCITT G.711 u-law. These codecs are tested in terms of its ability to deal with jitter buffer variations. The result shows that SJ Lab iLBC-20ms gives the best performance in terms of jitter buffer variation on LAN environment while SJ Lab GSM 6.10 shows the highest performance on wireless networks testing.




Jurnal Teknik Elektro Universitas Kristen Petra

Jurnal Teknik Elektro Universitas Kristen Petra is published biannually, in May and September, by... see more