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.
