On 29/04/11 3:25 AM, 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?
I am appreciate experienced inputs.
The text from that link:
Packet loss
Random packet loss is specified in the 'tc' command in percent. The
smallest possible non-zero value is:
232 = 0.0000000232%
# tc qdisc change dev eth0 root netem loss 0.1%
This causes 1/10th of a percent (i.e 1 out of 1000) packets to be
randomly dropped.
An optional correlation may also be added. This causes the random number
generator to be less random and can be used to emulate packet burst losses.
# tc qdisc change dev eth0 root netem loss 0.3% 25%
This will cause 0.3% of packets to be lost, and each successive
probability depends by a quarter on the last one.
Probn = .25 * Probn-1 + .75 * Random
--
Cheers,
Matt Riddell
_______________________________________________
http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/exchange.php (Full ITSP Solution)
http://www.venturevoip.com/cc.php (Call Centre Solutions)
--
_____________________________________________________________________
-- 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