Feature Requests item #1691183, was opened at 2007-03-30 11:22
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Category: None
Group: None
Status: Open
Resolution: None
Priority: 5
Private: No
Submitted By: MarcoMaz (marcomaz)
Assigned to: Nobody/Anonymous (nobody)
Summary: SIPp random traffic

Initial Comment:
Hi all.
For studying reasons I’ve built up a little LAN in my home, with two UA Pc and 
a third Pc with OpenSER installed on it.
I’d want to simulate a SIP net with 100 users, in which each of the users is 
able to speak to all the others, through SIPp call flows from UAC, via SER, to 
UAS.
Obviously the first thing I do is to register each of the users to SER with 
bindings to the IPs of both UAC and UAS Pcs, so that they can do and receive 
invitations.
Now I’m in trouble cause I don’t find a way to lounch SIPp so that the call 
flows are from each of “user_001”, “user_002”, etc, to each other, in random 
mode.
Any script or suggestion about that?

And, once the first problem is solved, I must find a way to avoid contemporary 
calls by the same user (i.e.: if “user_002” has been contacted by another user, 
than “user_002” must not be able to send an INVITE till the first call ends, 
and if “user_023” has invited another user, than “user_023” must not been able 
to send OK for another invitation).

Thank a lot.
Marco.


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