Abstract-In VoIP applications, packet loss causes a major impact on perceived speech quality. This impact is affected by some factors like packet loss locations, loss size and loss pattern. In this paper, we have investigated using perceptual-based objective measurement methods the impact on loss location on perceived speech quality and the relationships between convergence time and loss location for two different codecs (G.729 and G.723). Experimental result shows that loss location has a severe effect on perceived speech quality. The convergence time rely on the speech content e.g. voiced/unvoiced. In terms of unvoiced segments, the convergence time is stable as in voiced phases it varies. But it associate degree bound at the top of the segment. Our method allows a more accurate measurement of the exact effect of packet loss on perceived speech quality. As most of the internet subscribers of Asian countries use very low internet bandwidth. Hence, the goal of this analysis is to propose to change some parameters of this system so that the quality of voice may be kept in a tolerable limit using only 5kbps to 5.5kbps where this codec uses at least 6.4kbps. In a real life environment it is tested practically in research that it is possible to transmit voice satisfactorily using 5.3 kbps by changing some parameter described in this paper.Index Terms-G.723, G.729, annexb = no, RTP, UDP.
I. INTRODUCTIONOnce a call has been set up between two or more VoIP devices, the caller starts speaking. At this point the voice signal has to be converted into a digital signal, formatted for TCP/IP transmission and sent along the network to the destination, where all of the preceding steps have to be reversed. PCM produces a 64 kbps stream of data with excellent voice quality. This process allowed long distance calls to be places on the T1 lines of the telephone company for transmission. One voice call takes up one channel, not a very efficient scheme. With VoIP, we want to cram as much voice data into as little digital signal as possible. And instead of diverting our digital voice signal directly onto a T1 line, we need to packetize it and send it over an IP network.Encoding and compression techniques are published as standards by the International Telephone Union. Expect to see these when looking at specifications for VoIP equipment. Jitter buffers are memory areas used to store voice packets Manuscript received November 12, 2013; revised January 14, 2014 arriving with variable delays so that it appears that each voice sample has arrived in the same amount of time. The steady output of the voice samples from the jitter buffer is called playout. The playout is steady and constant, and as long as the jitter buffer receives an ample supply of voice packets, the system appears to have a fixed delay. VoIP is inefficient for small voice packets while large voice packets lead to long delays. The VoIP packet will have overhead in the form of headers. The headers for IP, UDP and RTP add up to 40 bytes. If the data was as small as 40 bytes...