How to test communicator (video, sound, microphone) implemented by using WebRTC

On my current job, we are developing an application that uses WebRTC technology.

And we want to test work of our application with 30 users in real time -- conference call with video, sound and microphone (and everything must work). We know that we can do it by real users (real users connected to our application).

Question is: How to test our requirement if we don't have such number of real persons? Maybe exists some tool for that.

Thanks.