[asterisk-users] Fw: Stress testing Asterisk

[Robert-GMAIL]

I believe there are options for rtp / audio..

Look at pcap play and rtp echo...

Transcoding would be another beast - if you are allowing it

Sent from my iPhone 5

On May 22, 2013, at 10:02 AM, Tommy Cooper <tomcooper83 at yahoo.com> wrote:

> From the little experience I have I do not think that that is a good way of testing the quality of voice. SIP only initiates and eventually terminates the call, once that the call is connected, SIP and therefore Asterisk are no longer involved. Once the call is connected it is assigned to a trapsport layer protocol such as RTP. RTP is the actual protocol that delivers the voice call between endpoints. I  believe that the setup of your network, QoS, codecs etc... determine the voice quality of your system.
> 
>  
> ----- Forwarded Message -----
> From: Mitul Limbani <mitul at enterux.in>
> To: Tommy Cooper <tomcooper83 at yahoo.com>; Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> 
> Sent: Wednesday, May 22, 2013 3:23 PM
> Subject: Re: [asterisk-users] Stress testing Asterisk
> 
> 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>
> To: Asterisk Users Mailing List - Non-Commercial Discussion <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> 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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> Mitul Limbani,
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