Abstract
The internet has revolutionized the telecommunication systems by supporting new applications and services. Voice over Internet Protocol (VoIP) is one of the most prominent telecommunication services based on IP. Asterisk is a popular VoIP services programs. The Asterisk supports many of audio codecs. The paper describes the VoIP based SIP, which is built by Asterisk. And it analyzes the bandwidth of G.711, G.726-32 and GSM. From the real traffic flow of audio data, it gets the radio of bandwidth. It provides valuable assessment to design the campus network's VoIP.
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