Voice Quality Assessment for Mobile to SIP Call over Live 3G Network |
The purpose of this paper is to assess the voice quality for mobile to SIP call on the live 3G
Venkatakrishnan G, Mkwawa IH, Sun L
network and to investigate the effects of codec and packet losses on the perceived speech
quality. Asterisk based test platform is used with SIP client on one end and connected to 3G
network on the other end to measure the speech quality on the live environment. More than
200 voice recordings are measured in the designed test bed on different codec combinations of
GSM-GSM and G711-GSM (codec from SIP to asterisk – codec from asterisk to mobile
phone) and results are analysed. Packet losses are introduced in the network to analyze the
impact on the speech quality on the live network. The result shown that GSM-GSM codec had
more impact on the speech quality with the MOS scores less than 3 whereas the G711-GSM
had a fair quality with MOS scores above 3 for most cases. Packet loss is found to have major
impact on voice quality and minor impact on call signalling on all the calls established with
the duration of 180 seconds or lesser. A formula is derived to predict the MOS values on
different packet loss conditions and validation tests on the proposed formula shown good
accuracy with the prediction errors range between ±0.3 MOS for most cases. The work should
help to better understand the voice quality for new services such as from 3G mobile to SIP
call.