|dc.description.abstract||In an age of fast-developing Internet with the advantage of low communication costof VoIP and improvement in communication quality, VoIP has been widely applied in
business or personal environment. However, the VoIP application software package like SKYPE has to be installed in advance in the PC at the customer site before using;besides, the complicated installation and configuration steps may beset the users who are not familiar to the computer operation; it will cause some trouble in usage. In the same time, for the computers in a public environment, pre-installation of software packages is not that convenient, this also leads to usage inconvenience. What could be more serious is that VoIP adopts SIP standard protocol; the said protocol does not offer sound solution to the problem of NAT penetration regarding the mutual call set-up for the communication which adopts RTP (Real Time Protocol), it thus leads to single-way voice communication situation or mutual communication can not be set up.
In this study, the author has focused on the above-mentioned problems and proposed a solution of integrating the convenience and popularity of the web browsing operation as well as the open feature of IAX2 protocol, and, developed a Web Phone Interface (It is named WebCall System). It frees the user from the perplexity and burden of the complicated installation procedures. Besides, in IAX2, both the signaling control message and media stream use the same network communication port to transmit, it could ameliorate the problems caused by NAT penetration in SIP. In order to avoid the possibility of impacting the communication quality due to missing too many packets
under the network environment of high error rate, in this study, we have made use of redundant packet protection technology to enhance the transmission accuracy and
selected Royalty-free and bandwidth-saving of GSM Codec Voice Compression format to alleviate the load of voice packet transmission. We have found out through the experiment that when using the Mean Opinion Score and the R-Factor Voice Quality Figures for measurement, under the environment of packet delay of 100 ms and packet
loss of 10%, the WebCall system demonstrates obvious 15% improvement comparing to the common VoIP system; it shows that an IAX2-based Webpage-Phone presents obvious improvement in bandwidth transmission efficiency in comparison to the traditional SIP Soft-Phone, this great helps the voice quality of mutual communication.