I guess technically you don’t have to make SIP phone calls to test RTP/SRTP. RTP is a separate protocol from SIP. But, I guess this really depends on what you’re wanting to test. Depending on how you have SIP setup in FreePBX, RTP will typically go directly from one phone to another and only connect to FreePBX/Asterisk for apps like voicemail and conferencing.
That being said, what are you trying to test? How many RTP streams a network can support? How many RTP streams a FreePBX box can take in?
You could use a trunk, looping the call several times. For example, set up a DID routed to an IVR with an option that (via a Misc Destination) calls out to the same number. You can then enter that option as many times as desired for the number of calls.
I don’t know whether a loopback trunk will provide a sufficiently accurate test, though you could try that first then confirm your results with a real PSTN trunk.
Note that there are many factors affecting PBX performance. By far, the biggest one is transcoding, especially with Opus. Call recording comes next, followed by encryption and listening for inband DTMF.
Also, depending on your endpoints, trunks, routers, etc., it may be possible to set up Asterisk for Direct Media, so the (S)RTP passes directly between device and trunk and the Asterisk load is zero, whether or not encryption is used. Of course, this also requires that Asterisk not transcode, record the call, or listen for DTMF.
Yes, exactly I want to test how many calls a Raspberry Pi with Asterisk can support. Now, I have switched off direct media.
The experiment is intended to answer how much more performance is required for encryption in a telephone system.
But now I’m not sure if I want to do it anymore.
However, I would like to do an experiment on RTP / SRTP / TLS /DTLS