Performance of Various Codecs Related to Jitter Buffer Variation in VoIP Using SIP
ABSTRACT: 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.
Author: Iwan Handoyo Putro
Journal Code: jptlisetrogg080001
