SIP Call Simulating

Hello everyone!

I want to measure performance differences between RTP and SRTP. To do this, I would have to simulate phone calls.

What programs or scripts are there?
As a tip I have already received SIPP, unfortunately I don’t quite understand it.
I use Asterisk 16 with FreePBX 15 on a Raspberry Pi 4.

As an alternative, I thought of opening a softphone several times to simulate conversations. Unfortunately I have not found one that can be opened multiple times and also allows SRTP.

There’s pjsua as well from pjsip.

1 Like

You can use Asterisk to simulate calls. SIPP is more useful for testing specific sequences of opertions.

Oh thanks, that looks good =)

What do you mean exactly?

  1. Create a ‘call file’ to ask Asterisk A to call Asterisk B.

  2. Push a bunch of call files into your outbound spool directory. A script to create the call file and ‘mv’ it is a good idea.

This works but not an efficient method, better use Originate command wit PHP or Python in a Loop

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.