<div dir="ltr">Here you go, George. I appreciate your energy into this: <a href="https://gist.github.com/ticklingcontest/4762a57457b73db1a170">https://gist.github.com/ticklingcontest/4762a57457b73db1a170</a></div><div class="gmail_extra"><br><div class="gmail_quote">On Sun, Mar 20, 2016 at 10:15 AM, George Joseph <span dir="ltr"><<a href="mailto:george.joseph@fairview5.com" target="_blank">george.joseph@fairview5.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div class="gmail_default" style="font-family:arial narrow,sans-serif"><br></div><div class="gmail_extra"><br><div class="gmail_quote"><span class="">On Sat, Mar 19, 2016 at 8:29 PM, Tickling Contest <span dir="ltr"><<a href="mailto:tickling.contest@gmail.com" target="_blank">tickling.contest@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Thanks again, George,<div><br></div><div>I am running PJSIP v. 2.4.5 that is prescribed for Asterisk 13.7.2. I made the changes you proposed re: --enable-epoll, recompiled PJSIP/Asterisk and redid the tests.</div><div><br></div><div>I am attempting a total of 100 or so connections (50 SIP/TCP callers REGISTERing, INVITEing; 50 SIP/TCP callees REGISTERing, accepting calls). I am afraid the latest Asterisk is not something I can try right now.</div><div><br></div></div></blockquote><div><br></div></span><div><div class="gmail_default" style="font-family:'arial narrow',sans-serif">How many sipp instance are you running?</div><div class="gmail_default" style="font-family:'arial narrow',sans-serif">Can you share the command lines and scenario xml files?</div><div class="gmail_default" style="font-family:'arial narrow',sans-serif"><br></div><br></div><div><div class="h5"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>I am left now with an issue that I have not been able to get help elsewhere re: SIPp. I hope I can ask them here, but if I can't, I am sorry in advance!</div><div><br></div><div>I get these errors:</div><div><br></div><div>Unable to bind audio RTP socket (IP=127.0.0.1, port=6100), errno = 98 (Address already in use). (also get the same error but with video RTP socket, but I am NOT running any video tests).<br></div><div><br></div><div>OR</div><div><br></div><div>Unable to bind main socket, errno = 98 (Address already in use).<br></div><div><br></div><div>OR</div><div><br></div><div><div>Unable to bind remote control socket (tried UDP ports 8888-8947): Address already in use. (no idea what this is, and why this occurs)</div></div><div><br></div><div>I tried passing the -mp parameter (for sipp, i.e.,) a random port number (e.g., see <a href="http://stackoverflow.com/questions/2556190/random-number-from-a-range-in-a-bash-script" target="_blank">http://stackoverflow.com/questions/2556190/random-number-from-a-range-in-a-bash-script</a>) but I still get these issues (though they dramatically decreased the number of issues I see due to port number).<br></div><div><br></div><div>Apart from ulimit/FD_SETSIZE related changes, what else can I do?</div><div><br></div><div>Where can I get information about load testing Asterisk for more than 100 concurrent calls (I use ARI, so I have to test my backend application too) etc.?</div><div><br></div><div>Thanks!</div></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Sat, Mar 19, 2016 at 5:13 PM, George Joseph <span dir="ltr"><<a href="mailto:george.joseph@fairview5.com" target="_blank">george.joseph@fairview5.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div style="font-family:'arial narrow',sans-serif"><br></div><div class="gmail_extra"><br><div class="gmail_quote"><span>On Sat, Mar 19, 2016 at 1:30 PM, Tickling Contest <span dir="ltr"><<a href="mailto:tickling.contest@gmail.com" target="_blank">tickling.contest@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr">Thank you, George!<div><br></div><div>That did solve the trouble for the most part, but I still get a few of these (not rarely, unfortunately):<div><div><br></div><div><div>Mar 19 15:23:06] ERROR[1981]: pjsip:0 <?>: <span style="white-space:pre-wrap"> </span>tcpc0x7fb0083a TCP connect() error: Connection refused [code=120111]</div><div>[Mar 19 15:23:06] WARNING[1981]: pjsip:0 <?>: <span style="white-space:pre-wrap"> </span>tsx0x7fb00008c Failed to send Request msg BYE/cseq=8737 (tdta0x7fb010246eb0)! err=120111 (Connection refused)</div><div>[Mar 19 15:23:16] ERROR[1981]: pjsip:0 <?>: <span style="white-space:pre-wrap"> </span>tcpc0x7fb0083a TCP connect() error: Connection refused [code=120111]</div><div>[Mar 19 15:23:16] WARNING[1981]: pjsip:0 <?>: <span style="white-space:pre-wrap"> </span>tsx0x7fb010169 Failed to send Request msg BYE/cseq=5278 (tdta0x7fb010096950)! err=120111 (Connection refused)</div></div><div><br></div><div>I set the value of PJ_IOQUEUE_MAX_HANDLES to FD_SETSIZE per the link you mentioned and ulimit for the root account, which runs asterisk is 8192.</div></div><div><br></div><div>The Asterisk VM has been upped to 8GB memory and 4 cores now. There cannot be network related latencies as the entire test is in the local network.<br></div><div><br></div></div><div></div></div></blockquote><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr"><div>Any insight is deeply appreciated.<br></div></div></blockquote><div><br></div><div><br></div></span><div><div style="font-family:'arial narrow',sans-serif;display:inline">T</div>hose errors you're seeing are outgoing connection attempts<div style="font-family:'arial narrow',sans-serif;display:inline"> so there a few things to check...</div> </div><div><br></div><div><span style="font-family:'arial narrow',sans-serif">What version of pjproject are you running? There's an issue in 2.4.5 and earlier where TCP sockets aren't being reused<div style="font-family:'arial narrow',sans-serif;display:inline">. It's fixed in their trunk. Unfortunately, you can't use their trunk with Asterisk 13.7.2 because of a new api they introduced. You'll have to use Asterisk's current 13 branch from git. If you 're going to do that, check out the bundled pjproject option which also has that patch.</div></span><br></div><div><span style="font-family:'arial narrow',sans-serif"><br></span></div><div><div style="font-family:'arial narrow',sans-serif;display:inline">Check that sipp isn't terminating early and forcing Asterisk to open a new connection just to send the BYE.</div><div style="font-family:'arial narrow',sans-serif;display:inline"></div><br></div><div><br></div><div><div style="font-family:'arial narrow',sans-serif">How many connections are you attempting?<br></div></div><div style="font-family:'arial narrow',sans-serif"><br></div><div style="font-family:'arial narrow',sans-serif">If you really want to get the max connections, set the --enable-epoll option on pjproject's ./configure line (assuming you're on Linux) and set <span style="font-family:arial,sans-serif">PJ_IOQUEUE_MAX_HANDLES to 5000 or something. This won't fix the connection refused messages however.</span></div><div><div><div style="font-family:'arial narrow',sans-serif"><br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Sat, Mar 19, 2016 at 12:09 PM, George Joseph <span dir="ltr"><<a href="mailto:george.joseph@fairview5.com" target="_blank">george.joseph@fairview5.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr"><div style="font-family:'arial narrow',sans-serif">I'll bet your pjproject install still has the default value of 64 for PJ_IOQUEUE_MAX_HANDLES. That limits the number of simultaneous TCP sockets.</div><div style="font-family:'arial narrow',sans-serif"><br></div><div><font face="arial narrow, sans-serif"><a href="https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject</a></font><br></div><div><font face="arial narrow, sans-serif"><br></font></div><div><font face="arial narrow, sans-serif"><br></font></div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Sat, Mar 19, 2016 at 9:20 AM, Tickling Contest <span dir="ltr"><<a href="mailto:tickling.contest@gmail.com" target="_blank">tickling.contest@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div><div><div dir="ltr">I am load-testing an Asterisk 13.7.2 installation with SIPp 3.5.1.<div><br></div><div>I am running into an issue where Asterisk (core set debug 99) complains about the following:</div><div><br></div><div><div>[Mar 19 11:12:55] WARNING[13078]: pjsip:0 <?>: <span style="white-space:pre-wrap"> </span>tsx0x7fc60840e ...Failed to send Request msg INVITE/cseq=28144 (tdta0x7fc60840b590)! err=70010 (Too many objects of the specified type (PJ_ETOOMANY))</div><div>[Mar 19 11:12:55] WARNING[13078]: pjsip:0 <?>: <span style="white-space:pre-wrap"> </span>tsx0x7fc60841a ...Failed to send Request msg INVITE/cseq=31094 (tdta0x7fc6140ebc10)! err=70010 (Too many objects of the specified type (PJ_ETOOMANY))</div><div>[Mar 19 11:13:22] ERROR[13083]: pjsip:0 <?>: <span style="white-space:pre-wrap"> </span>tcpc0x7fc6082b TCP connect() error: Connection refused [code=120111]</div><div>[Mar 19 11:13:22] WARNING[13083]: pjsip:0 <?>: <span style="white-space:pre-wrap"> </span>tsx0x7fc60822e Failed to send Request msg BYE/cseq=29352 (tdta0x7fc6080e3600)! err=120111 (Connection refused)</div><div>[Mar 19 11:13:25] ERROR[13083]: pjsip:0 <?>: <span style="white-space:pre-wrap"> </span>tcpc0x7fc6083a TCP connect() error: Connection refused [code=120111]</div><div>[Mar 19 11:13:25] WARNING[13083]: pjsip:0 <?>: <span style="white-space:pre-wrap"> </span>tsx0x7fc60822e Failed to send Request msg BYE/cseq=13911 (tdta0x7fc6140bc200)! err=120111 (Connection refused)</div></div><div><br></div><div>Any help figuring out this issue is deeply appreciated.</div><div><br></div><div>Thanks!</div></div>
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