Hi guys! I did my best yesterday to be on the conference but it was impossible due to my connection.. My two cents. Tests should include full calls between the differents endpoints. Sip/Sip, Sip/WebRTC, WebRTC/Sip, WebRTC/WebRTC, udo, tcp, tls, tls/srtp(in case of sip) Sample config for the test should be included to inspect and study PjSip has a command line UA, that could be used to make calls, play some sort of pre recorded audio using some scripting. -- Víctor E. Medina M. Platform Architect / Chief Infrastructure +58424 291 4561 BB #79A8AFA2 @VMCibersys