<html><head><meta http-equiv="Content-Type" content="text/html; charset=UTF-8"></head><body><div>I'm not sure what a single instance of rtpproxy can handle, but most people squeezing thousand of concurrent calls per box are probably doing it on multicore boxes by binding multiple instances of rtpproxy with different core affinities, and round-robining among them. </div><div><br></div><div><br></div><div><br></div><div><br></div><div><div style="font-size:75%;color:#575757">-- Alex<br><br>--<br>Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. <br><br>Alex Balashov - Principal<br>Evariste Systems LLC<br>235 E Ponce de Leon Ave<br>Suite 106<br>Decatur, GA 30030<br>Tel: +1-678-954-0670<br>Web: http://www.evaristesys.com/</div></div> <br>Mino Haluz <mino.haluz@gmail.com> wrote:<br>The results:<br><br>- rtpproxy calls count 280<br>- sipp calls count 2000<br>- iptraf on proxy 4.8MB/s<br>- G711a codec<br><br>So if my calculations are right (16kB/s per stream * 280 = 4.5MB/s),<br>rtpproxy calls count is really the right value. CPU usage is ok on<br>every machine (rtpproxy 20-30% CPU). Does anybody know why rtpproxy<br>cannot serve more than 270-280 calls ?<br><br>On Thu, Sep 13, 2012 at 5:07 PM, Mino Haluz <mino.haluz@gmail.com> wrote:<br>> Ok, so I put there unforce_rtp_proxy even though I'm using<br>> rtpproxy_manage. The tip with nc now really shows the calls count.<br>><br>> But the dialog count is still higher and higher, so I have bug<br>> somewhere in the configuration. I'll check it.<br>><br>> On Thu, Sep 13, 2012 at 4:53 PM, Alex Balashov<br>> <abalashov@evaristesys.com> wrote:<br>>> Correct, but you still need to call rtpproxy_manage() on receipt of a BYE or<br>>> CANCEL. It'll just figure out what to do on its own.<br>>><br>>> None of this has to do with dialog state, though. Just rtpproxy control.<br>>><br>>><br>>><br>>><br>>> -- Alex<br>>><br>>> --<br>>> Sent from my Samsung mobile, and thus lacking in the refinement one might<br>>> expect from a proper keyboard.<br>>><br>>> Alex Balashov - Principal<br>>> Evariste Systems LLC<br>>> 235 E Ponce de Leon Ave<br>>> Suite 106<br>>> Decatur, GA 30030<br>>> Tel: +1-678-954-0670<br>>> Web: http://www.evaristesys.com/<br>>><br>>> Mino Haluz <mino.haluz@gmail.com> wrote:<br>>> I'm using rtpproxy_manage, so I assume unforce_rtp is not needed.<br>>><br>>> On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov <lemenkov@gmail.com> wrote:<br>>>> 2012/9/13 Mino Haluz <mino.haluz@gmail.com>:<br>>>><br>>>>> Peter: Thanks for the tip! Really interesting. But I do not<br>>>>> understand, why also this list contains the calls that were ended by<br>>>>> sipp... Should I search for some mistake in my kamaillio config ?<br>>>><br>>>> Perhaps you don't close them with unforce_rtp_proxy:<br>>>><br>>>> if(method=="BYE" || method=="CANCEL"){<br>>>> unforce_rtp_proxy();<br>>>> }<br>>>><br>>>> --<br>>>> With best regards, Peter Lemenkov.<br>>>><br>>>> _______________________________________________<br>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>>>> sr-users@lists.sip-router.org<br>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users<br>>><br>>> _______________________________________________<br>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>>> sr-users@lists.sip-router.org<br>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users<br>>><br>>> _______________________________________________<br>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>>> sr-users@lists.sip-router.org<br>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users<br>>><br></body>