>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 <mi...@enterux.in>
To: Tommy Cooper <tomcoope...@yahoo.com>; Asterisk Users Mailing List -
Non-Commercial Discussion <asterisk-users@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 <ma...@vtl.ee>
>To: Asterisk Users Mailing List - Non-Commercial Discussion
><asterisk-users@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 <tomcoope...@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.
>
>
>
>--
>_____________________________________________________________________
>-- 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
>
>
>
--
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: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422
--
_____________________________________________________________________
-- 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
