asterisk users - Apr 2011 - How to create distortion, echo, and chopping sound in a SIP trunk?

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.

Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20110428/0d2edf56/attachment.htm>

Hey,

Not sure why you would want to do this. I find nothing destroys a clean 
link like running torrents. Try downloading the 10 most popular torrents 
off of thepiratebay.org ( that are more than 4 gigs ).

( just make sure aren't breaking any copyright rules ).

That'll saturate your link and will give you lots of distortions on your 
VoIP.

David


On 2011-04-28 11:25, 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.
>
> Thanks
>
>
> --
> _____________________________________________________________________
> -- 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20110428/180834bc/attachment.htm>

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 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 198 bytes
Desc: This is a digitally signed message part.
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20110428/313be855/attachment.pgp>

In article <BANLkTim8W+vjJJ87oYy1MVpPsfwfLUt0Kg at mail.gmail.com>,
Bruce B <bruceb444 at gmail.com> 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.
You could use iptables to cause random packet loss.

See
http://code.nomad-labs.com/2010/03/11/simulating-dropped-packets-aka-crappy-internets-with-iptables/
for examples. You might want to precede those rules with ACCEPT rules
for the traffic you want to remain reliable (such as TCP connections).

Cheers
Tony
-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org

You can use tc (traffic control) on linux and limit your bandwidth

http://www.linuxtoday.com/infrastructure/2008092400820OSDBNT 

> To: asterisk-users at lists.digium.com
> From: tony at softins.co.uk
> Date: Thu, 28 Apr 2011 15:57:26 +0000
> Subject: Re: [asterisk-users] How to create distortion, echo,	and chopping
sound in a SIP trunk?
> 
> In article <BANLkTim8W+vjJJ87oYy1MVpPsfwfLUt0Kg at mail.gmail.com>,
> Bruce B <bruceb444 at gmail.com> 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.
> 
> You could use iptables to cause random packet loss.
> 
> See
http://code.nomad-labs.com/2010/03/11/simulating-dropped-packets-aka-crappy-internets-with-iptables/
> for examples. You might want to precede those rules with ACCEPT rules
> for the traffic you want to remain reliable (such as TCP connections).
> 
> Cheers
> Tony
> -- 
> Tony Mountifield
> Work: tony at softins.co.uk - http://www.softins.co.uk
> Play: tony at mountifield.org - http://tony.mountifield.org
> 
> --
> _____________________________________________________________________
> -- 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
 		 	   		  
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20110428/97568e59/attachment.htm>

On 4/28/11 5:25 PM, Bruce B wrote:
> Is there any easy way to simulate a distorted SIP line temporarily for
> testing?
Build a Linux based router and use netem/tc to mess around with the
routed traffic. You can insert packetloss, jitter, etc and have it be
reproducable.

-- 
Andreas Sikkema

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.
Use the following link to simulate packet loss.

http://www.linuxfoundation.org/collaborate/workgroups/networking/netem#Packet_loss

-- 
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)

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)

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

2 maj 2011 kl. 18.09 skrev Hans Witvliet:
> 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
I use a laptop with a usb-ethernet connected in bridge mode as a "voip
destroyer".
Using TC you can inject a lot of bad stuff on the connection.

/O