Practical Analysis of Asterisk SIP Server Performance

This paper presents an analysis of the Asterisk server performance as a SIP server, as well as of bandwidth consumption in multiple scenarios. Server load has been tested using a certain type of codec, which requires a voice quality close to classic telephony (PSTN). Therefore, one can infer minimum VoIP requirements for an Asterisk SIP server by analyzing the maximum number of possible calls and the maximum bandwidth.