3 thoughts on - Stress Testing Asterisk

  • I have a question here.

    How can we test the quality of voice upon increasing the call load?

    Can we try passing a voice file using sipp and record the same in dial plan record application ? Is this reliable enough to simulate near real world scenario?


  • Easily, as long as you have no media 🙂

    Use -sn uac_pcap instead of -sn uac to test with RTP (and watch your call count drop). Add recording (MixMonitor()) to your dialplan and watch the call count go down even more. 😉

    A rough way to see if call quality is deteriorating would be to call your Asterisk box while the SIPP test is running and listen to some message played via Background().