<html><body><div style="color:#000; background-color:#fff; font-family:times new roman, new york, times, serif;font-size:12pt"><div style="RIGHT: auto"><SPAN style="RIGHT: auto">From the little experience I have I do not think that that is a good way of testing the quality of voice. SIP </SPAN><SPAN style="RIGHT: auto">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<VAR id=yui-ie-cursor></VAR>. I believe that the setup of your network, QoS, codecs etc... determine the voice quality of your system.</SPAN></div>
<div style="BACKGROUND-COLOR: transparent; FONT-STYLE: normal; FONT-FAMILY: times new roman, new york, times, serif; COLOR: rgb(0,0,0); FONT-SIZE: 16px; RIGHT: auto"><SPAN style="RIGHT: auto"></SPAN><BR style="RIGHT: auto"> </div>
<DIV style="FONT-FAMILY: times new roman, new york, times, serif; FONT-SIZE: 12pt">
<DIV style="FONT-FAMILY: times new roman, new york, times, serif; FONT-SIZE: 12pt">
<DIV style="RIGHT: auto" dir=ltr>----- Forwarded Message -----<BR><FONT size=2 face=Arial><B><SPAN style="FONT-WEIGHT: bold">From:</SPAN></B> Mitul Limbani <mitul@enterux.in><BR><B><SPAN style="FONT-WEIGHT: bold">To:</SPAN></B> Tommy Cooper <tomcooper83@yahoo.com>; Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> <BR><B><SPAN style="FONT-WEIGHT: bold">Sent:</SPAN></B> Wednesday, May 22, 2013 3:23 PM<BR><B><SPAN style="FONT-WEIGHT: bold">Subject:</SPAN></B> Re: [asterisk-users] Stress testing Asterisk<BR></FONT></DIV>
<DIV class=y_msg_container><BR>
<DIV id=yiv0649989030>I have a question here.
<DIV><BR></DIV>
<DIV>How can we test the quality of voice upon increasing the call load?</DIV>
<DIV><BR></DIV>
<DIV>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?</DIV>
<DIV><BR></DIV>
<DIV>Mitul<SPAN></SPAN><BR><BR>On Wednesday, May 22, 2013, Tommy Cooper wrote:<BR>
<BLOCKQUOTE style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class=yiv0649989030gmail_quote>
<DIV>
<DIV style="FONT-FAMILY: times new roman, new york, times, serif; FONT-SIZE: 12pt">
<DIV><SPAN>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?<VAR></VAR><BR></SPAN></DIV>
<DIV><BR></DIV>
<DIV style="FONT-FAMILY: times new roman, new york, times, serif; FONT-SIZE: 12pt">
<DIV style="FONT-FAMILY: times new roman, new york, times, serif; FONT-SIZE: 12pt">
<DIV dir=ltr>----- Forwarded Message -----<BR><FONT face=Arial><B><SPAN style="FONT-WEIGHT: bold">From:</SPAN></B> Marie Fischer <<A href="" rel=nofollow>marie@vtl.ee</A>><BR><B><SPAN style="FONT-WEIGHT: bold">To:</SPAN></B> Asterisk Users Mailing List - Non-Commercial Discussion <<A href="" rel=nofollow>asterisk-users@lists.digium.com</A>> <BR><B><SPAN style="FONT-WEIGHT: bold">Sent:</SPAN></B> Wednesday, May 22, 2013 1:16 PM<BR><B><SPAN style="FONT-WEIGHT: bold">Subject:</SPAN></B> Re: [asterisk-users] Stress testing Asterisk<BR></FONT></DIV>
<DIV><BR><BR>On 21.05.2013, at 0:05, Tommy Cooper <<A href="" rel=nofollow>tomcooper83@yahoo.com</A>> wrote:<BR><BR>> Hi,<BR>> 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.<BR><BR>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.<BR><A href="http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/" rel=nofollow target=_blank>http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/</A>has some simple instructions which should get you started.<BR>If the calls still fail, Asterisk console output would be helpful.<BR><BR><BR><BR>--<BR>_____________________________________________________________________<BR>-- Bandwidth
and Colocation Provided by <A href="http://www.api-digital.com/" rel=nofollow target=_blank>http://www.api-digital.com/</A>--<BR>New to Asterisk? Join us for a live introductory webinar every Thurs:<BR> <A href="http://www.asterisk.org/hello" rel=nofollow target=_blank>http://www.asterisk.org/hello</A><BR><BR>asterisk-users mailing list<BR>To UNSUBSCRIBE or update options visit:<BR> <A href="http://lists.digium.com/mailman/listinfo/asterisk-users" rel=nofollow target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR><BR><BR></DIV></DIV></DIV></DIV></DIV></BLOCKQUOTE></DIV><BR><BR>-- <BR>Regards,<BR>Mitul Limbani,<BR>Chief Architech & Founder,<BR>Enterux Solutions Pvt. Ltd.<BR>110 Reena Complex, Opp. Nathani Steel, <BR>Vidyavihar (W), Mumbai - 400 086. India<BR><A href="http://www.enterux.com/" rel=nofollow target=_blank>http://www.enterux.com/</A><BR><A
href="http://www.entvoice.com/" rel=nofollow target=_blank>http://www.entvoice.com/</A><BR>email: <A href="mailto:mitul@enterux.in" rel=nofollow target=_blank ymailto="mailto:mitul@enterux.in">mitul@enterux.in</A><BR>DID: +91-22-71967121<BR>Cell: +91-9820332422<BR><BR><BR></DIV><BR><BR></DIV></DIV></DIV></div></body></html>