From: SourceForge.net <no...@so...> - 2007-03-30 11:22:42
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Feature Requests item #1691183, was opened at 2007-03-30 11:22 Message generated for change (Tracker Item Submitted) made by Item Submitter You can respond by visiting: https://sourceforge.net/tracker/?func=detail&atid=637567&aid=1691183&group_id=104305 Please note that this message will contain a full copy of the comment thread, including the initial issue submission, for this request, not just the latest update. 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. ---------------------------------------------------------------------- You can respond by visiting: https://sourceforge.net/tracker/?func=detail&atid=637567&aid=1691183&group_id=104305 |