[asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?

[Stefan Gofferje]

On Thursday 28 April 2011, Bruce B wrote:
 
> 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?
> 
> I am appreciate experienced inputs.

Force the switch port which the asterisk is connected to 10MBit/s half-duplex 
and then fire a ping -f -s 65507 <asterisk-host> from a machine with a 
gigabit-link to the switch.
That should get the line quality pretty much to the bottom.

-S

-- 
 (o_   Stefan Gofferje            | SCLT, MCP, CCSA
 //\   Reg'd Linux User #247167   | VCP #2263
 V_/_  Heckler & Koch - the original point and click interface 
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