[asterisk-users] Stress testing Asterisk

Mitul Limbani mitul at enterux.in
Wed May 22 08:23:26 CDT 2013


I have a question here.

How can we test the quality of voice upon increasing the call load?

Can we try passing a voice file using sipp and record the same in dial plan
record application ? Is this reliable enough to simulate near real world
scenario?

Mitul

On Wednesday, May 22, 2013, Tommy Cooper wrote:

> Thank you for your help I finally solved this issue. Is it possible that
> my setup can achieve 212 concurrent calls, I am running Asterisk on just 1
> core using 3.5 GHz, and 1Gb of RAM?
>
>  ----- Forwarded Message -----
> *From:* Marie Fischer <marie at vtl.ee <javascript:_e({}, 'cvml',
> 'marie at vtl.ee');>>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users at lists.digium.com <javascript:_e({}, 'cvml',
> 'asterisk-users at lists.digium.com');>>
> *Sent:* Wednesday, May 22, 2013 1:16 PM
> *Subject:* Re: [asterisk-users] Stress testing Asterisk
>
>
> On 21.05.2013, at 0:05, Tommy Cooper <tomcooper83 at yahoo.com<javascript:_e({}, 'cvml', 'tomcooper83 at yahoo.com');>>
> wrote:
>
> > Hi,
> > I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is
> generating are failing. I am trying to run Sipp on the same machine as
> Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command.
>
> Do you have a peer and extension configured for SIPP in your Asterisk
> configuration? You also needat least the -s <extension_to_dial> option on
> your sipp command line.
>
> http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/has
> some simple instructions which should get you started.
> If the calls still fail, Asterisk console output would be helpful.
>
>
>
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-- 
Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mitul at enterux.in
DID: +91-22-71967121
Cell: +91-9820332422
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