<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div>I believe there are options for rtp / audio..</div><div><br></div><div>Look at pcap play and rtp echo...</div><div><br></div><div>Transcoding would be another beast - if you are allowing it<br><br>Sent from my iPhone 5</div><div><br>On May 22, 2013, at 10:02 AM, Tommy Cooper <<a href="mailto:tomcooper83@yahoo.com">tomcooper83@yahoo.com</a>> wrote:<br><br></div><blockquote type="cite"><div><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 <<a href="mailto:mitul@enterux.in">mitul@enterux.in</a>><br><b><span style="FONT-WEIGHT: bold">To:</span></b> Tommy Cooper <<a href="mailto:tomcooper83@yahoo.com">tomcooper83@yahoo.com</a>>; Asterisk Users Mailing List - Non-Commercial Discussion <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>> <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></div></blockquote><blockquote type="cite"><div><span>--</span><br><span>_____________________________________________________________________</span><br><span>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --</span><br><span>New to Asterisk? Join us for a live introductory webinar every Thurs:</span><br><span> <a href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a></span><br><span></span><br><span>asterisk-users mailing list</span><br><span>To UNSUBSCRIBE or update options visit:</span><br><span> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></span></div></blockquote></body></html>