On Thu, 2011-04-28 at 11:25 -0400, Bruce B wrote: > Hi everyone, > > > How can I introduce some distortion, echo, chopping sound and all > other bad quality things that can happen to a SIP trunk? I have plenty > of bandwidth and crisp clear lines so the only thing that I can think > of is to limit bandwidth but even that requires quite some scripting > work. > > > Is there any easy way to simulate a distorted SIP line temporarily for > testing?
You can intruduce a predefined amount of "distortion" on your ip-connection (packet loss, fluctuating delay, out of secuence reception of packets, limited bandwith) All of these will have a serious impact on your VOIP-connection. See "lartc" about it. Good thing about it, is that you pre-define how bad a line is, and it produces re-producable results hw -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
