I am writing this to let you know that this behavior persists in the 1.0.4pre9.
Could the calls/sec issue be due to the single threaded nature of Sofia? Because I am getting the feeling that the number of simultaneous channels doesn't really burdens FS, but many Calls/sec does. Apostolos Pantsiopoulos wrote: > Anthony Minessale wrote: >> FS uses async rtp timers so you may want to set rtp-timer-name=none in >> the profile param to simulate asterisk conditions. > > I tried that - although I am not using rtp in my scenario - with the > same results. > >> Also keep in mind that asterisk as an atvantage in a tiny crappy 32 bit >> single cpu box because that was what was popular when it was designed >> and the chance for race conditions is minimal because there is only 1 >> cpu. As you scale up to a 8 core 64 bit xeon you will set a drastic >> difference. > > Yes I know that this machine is not well suited for today's test needs. > But the issue occurs in every machine as long as it is pushed near (but > not quite near) to its limits. I have the same odd durations using a 64 > bit low end server. In this case I could achieve a better call/sec rate > than that of the crappy PC but around 50-60 calls/sec the same problem > showed up. I also used a Mosso Cloud Server (quad core - 64-bit) and the > same thing happened at a higher rate. > > >> I will be happy to investigate this issue a bit if you'd like but i do >> not have any box like you describe so if I can't find anything >> you may have to lend us your lab. > > I would appreciate it if you did. After all there this might be a > problem that has not surfaced yet but someday will as more and more > production boxes start using FS. So it would be better to investigate it > now. > I don't think lending you access to my old P4 PC would help you very much :) > If you have access to a normal 2-4 core system you can easily start > raising the sipp parameters until it starts happening. However if you > really think it is appropriate to use my test machines I'd be happy to > grant access to our low-end Opteron machine (just send me a personal > email). I cannot grant you access to larger systems because they are > used in production. > > I used the embedded sipp scenarios : > > on the UAS side : > > sipp -i <UAS_IP> -mi <UAS_IP> -ci <UAS_IP> -mp 8000 -sn uas > > on the UAC side : > > sipp <FS_IP>:5060 -s 44050505-i <UAC_IP> -mi <UAC_IP> -ci <UAC_IP> -r 70 > -d 5000 -l 500 -m 2000 -sn uac > > The dialplan : > > <?xml version="1.0" encoding="utf-8"?> > <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML --> > <include> > > <context name="mydialplan"> > <extension name="dial1"> > <condition field="destination_number" expression="(^.*)$"> > <!-- Dial Back --> > <action application="set" > data="absolute_codec_string=PCMU"/> > <!-- <action application="set" > data="proxy_media=true"/> --> > <action application="bridge" > data="sofia/gateway/sipp01/$1"/> > </condition> > </extension> > </context> > > </include> > > If you need anything else from the config just notify me. > > In order to verify that at some point the calls start having a > duration larger than the scenario's 5secs you can tcpdump on the sipp > machine or turn on the cdrs logging (I know that it degrades > performance, but as I said it is not a matter of when exactly it > starts happening, it is a matter that it DOES start happening). > > >> >> On Thu, Jun 4, 2009 at 12:47 PM, r...@kinetix.gr >> <mailto:r...@kinetix.gr> <r...@kinetix.gr <mailto:r...@kinetix.gr>> wrote: >> >> Michael Collins wrote: >> > >> > >> > The dialplan : >> > >> > <?xml version="1.0" encoding="utf-8"?> >> > <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML --> >> > <include> >> > >> > <context name="mydialplan"> >> > <extension name="dial1"> >> > <condition field="destination_number" >> > expression="^.*$"> >> > >> > >> > You forgot the parens around .* >> > It should be expression="^(.*)$" if you plan to use $1 later in the >> > extension... >> > >> > >> > >> > <!-- Dial Back --> >> > <action application="set" >> > data="absolute_codec_string=PCMA"/> >> > <action application="bridge" >> > data="sofia/gateway/sipp01/$1"/> >> > >> > ... like here ^^^^^^^ >> > :) >> > -MC >> >> You are right! Although, I don't think that would change the outcome of >> my test :) >> > >> > >> > >> > </condition> >> > </extension> >> > </context> >> > >> > </include> >> > >> > >> > >> ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users@lists.freeswitch.org >> <mailto:Freeswitch-users@lists.freeswitch.org> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> <mailto:Freeswitch-users@lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_miness...@hotmail.com >> <mailto:msn%3aanthony_miness...@hotmail.com> >> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com >> <mailto:paypal%3aanthony.miness...@gmail.com> >> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch >> >> FreeSWITCH Developer Conference >> sip:8...@conference.freeswitch.org >> <mailto:sip%3a...@conference.freeswitch.org> >> iax:gu...@conference.freeswitch.org/888 >> <http://iax:gu...@conference.freeswitch.org/888> >> googletalk:conf+...@conference.freeswitch.org >> <mailto:googletalk%3aconf%2b...@conference.freeswitch.org> >> pstn:213-799-1400 >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: r...@kinetix.gr ------------------------------------------- _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org