<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class=""><br class=""><div><blockquote type="cite" class=""><div class="">On Mar 2, 2016, at 11:30 AM, Tickling Contest <<a href="mailto:tickling.contest@gmail.com" class="">tickling.contest@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class="">Hello,<div class=""><br class=""></div><div class="">I have an ARI application that I would like to load test using SIPp (<a href="http://sipp.sourceforge.net/" class="">http://sipp.sourceforge.net/</a>).</div><div class=""><br class=""></div></div></div></blockquote><div><br class=""></div>Shameless plug: you may find SippyCup will help make creating load test profiles easier: <a href="https://mojolingo.github.com/sippy_cup" class="">https://mojolingo.github.com/sippy_cup</a></div><div><br class=""><blockquote type="cite" class=""><div class=""><div dir="ltr" class=""><div class="">I understand how to REGISTER and send an INVITE out to a callee, but it is not clear to me:</div><div class=""><br class=""></div><div class="">(a) how to choose a RANDOM port for each client I launch using say <a href="http://tomeko.net/other/sipp/sipp_cheatsheet.php?lang=pl" class="">http://tomeko.net/other/sipp/sipp_cheatsheet.php?lang=pl</a> </div><div class="">(REGISTER_INVITE_client.xml). Should I generate this beforehand and pass it as field3, say? Or is there a method already available in SIPp to do this?</div></div></div></blockquote><div><br class=""></div><div>I don’t know about making the ports random, but you can allocate a unique port-per-connect by passing the “-t un” flag to sipp. Did you really need them random, or just unique? And out of curiosity, why? Most of the time, the default of “-t u1”, which is a single UDP port shared for all dialogs, works fine.</div><br class=""><blockquote type="cite" class=""><div class=""><div dir="ltr" class=""><div class="">(b) How do I receive a call into a SIPp client? In other words, how do I simulate a SIPp client to REGISTER and then wait for a call (from a SIPp client connected to the same PBX)?</div></div></div></blockquote><div><br class=""></div><div>This gets tricky because SIPp isn’t a real UA. However, you can write a SIPp scenario that sends a REGISTER then waits for an INVITE. The Sippy Cup documentation has an example of this.</div><br class=""><blockquote type="cite" class=""><div class=""><div dir="ltr" class=""><div class="">(c) Can I play an audio file (or an mp3) after the call starts from both the caller and callee?</div></div></div></blockquote><div><br class=""></div>You can replay audio in the form of a pcap file. If I need real audio, the easiest path is usually to place a real call with tcpdump running.</div><div><a href="http://sipp.sourceforge.net/doc/reference.html#PCAP+Play" class="">http://sipp.sourceforge.net/doc/reference.html#PCAP+Play</a></div><div><br class=""></div><div><br class=""></div><div>/BAK/</div><div><br class=""></div><div><div class=""><div style="line-height: normal; orphans: 2; text-align: -webkit-auto; widows: 2; word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class=""><span class="Apple-style-span" style="border-collapse: separate; line-height: normal; border-spacing: 0px;"><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class=""><span class="Apple-style-span" style="border-collapse: separate; line-height: normal; text-align: -webkit-auto; border-spacing: 0px;"><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class=""><span class="Apple-style-span" style="border-collapse: separate; line-height: normal; text-align: -webkit-auto; border-spacing: 0px;"><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class=""><span class="Apple-style-span" style="border-collapse: separate; line-height: normal; border-spacing: 0px;"><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class=""><div class=""><div class="">-- </div><div class="">Ben Klang</div><div class="">Principal/Technology Strategist, Mojo Lingo</div><div class=""><a href="mailto:bklang@mojolingo.com" class="">bklang@mojolingo.com</a></div><div class="">+1.404.475.4841</div><div class=""><br class=""></div><div class="">Mojo Lingo -- <i class="">Voice applications that work like magic</i></div><div class=""><a href="http://mojolingo.com/" class="">http://mojolingo.com</a></div></div><div class="">Twitter: @MojoLingo</div><div class=""><br class=""></div></div></span></div></span></div></span></div></span></div></div><div class=""><br class="webkit-block-placeholder"></div><blockquote type="cite" class=""><div class=""><div dir="ltr" class=""><div class=""><br class=""></div><div class="">Your help is received with thanks!<br class=""></div></div>
_______________________________________________<br class="">asterisk-app-dev mailing list<br class=""><a href="mailto:asterisk-app-dev@lists.digium.com" class="">asterisk-app-dev@lists.digium.com</a><br class="">http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev<br class=""></div></blockquote></div><br class=""></body></html>