Sipp's media stack doesn't seem to be very good. what i did was to use freeswitch. You can generate calls to fs (or make a script to "originate" staright from fs, though sipp is simples), which will resend it to kamailio. And on answer, it will send the call to a callcenter that will just playback music on hold. From that moment on, the call belongs to freeswitch.
That worked fine for us.
Regards,
David On Sun, Oct 2, 2016 at 10:27 AM Gholamreza Sabery gr.sabery@gmail.com wrote:
I used SIPp to stress test different scenarios in Kamailio, but how can I simulate a real call with media? SIPp has media sending capabilities but in not enough for example to simulate 1000 calls persecond! How can I test this? _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users