14 feb 2008 kl. 21.25 skrev SVN commits to the Digium repositories: > Author: murf > Date: Thu Feb 14 14:25:11 2008 > New Revision: 103686 > > URL: http://svn.digium.com/view/asterisk?view=rev&rev=103686 > Log: > Adding some documentation for testing sip channel performance using > sipp.
Murf! Thank you! Maybe we can start creating a collection of SIPP tests to run various scenarious. I would like to test how registrations and subscriptions affect the stack too. Imagine your PVT list with 500 subscriptions, 50 per extension. Then place 200 calls and you will see asterisk having a lot of fun delivering call states to 50 subscribers per extension. That would be a good stress test for your branch, since it not only involves INVITE/BYE but also traversing the list for SUBSCRIBEs and having a much larger list of PVT's than just the calls. /O _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
