Hi Anthony, I'm aware it is generating 30 retries per a call and this is killing me ...
I lost my entire working day to figure out what is missing in the damn ACK message SIPp is sending back... ACK looks quite ok to me. pls can you help ? freeswi...@l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at 16:44:26.527236: ------------------------------------------------------------------------ INVITE sip:30003016094191...@10.4.4.251<sip%3a30003016094191...@10.4.4.251>SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport Max-Forwards: 70 Contact: <sip:22222238515000...@10.4.4.252<sip%3a22222238515000...@10.4.4.252> > To: "30003016094191500"<sip:30003016094191...@10.4.4.251<sip%3a30003016094191...@10.4.4.251> > From: "22222238515000403"<sip:22222238515000...@10.4.4.251<sip%3a22222238515000...@10.4.4.251> >;tag=1 Call-ID: 1-6...@10.4.4.252 CSeq: 1 INVITE Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 131 v=0 o=user1 53655765 2353687637 IN IP4 10.4.4.252 s=- c=IN IP4 10.4.4.252 t=0 0 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ------------------------------------------------------------------------ send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060 From: "22222238515000403"<sip:22222238515000...@10.4.4.251<sip%3a22222238515000...@10.4.4.251> >;tag=1 To: "30003016094191500"<sip:30003016094191...@10.4.4.251<sip%3a30003016094191...@10.4.4.251> > Call-ID: 1-6...@10.4.4.252 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582: ------------------------------------------------------------------------ SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060 From: "22222238515000403"<sip:22222238515000...@10.4.4.251<sip%3a22222238515000...@10.4.4.251> >;tag=1 To: "30003016094191500" <sip:30003016094191...@10.4.4.251<sip%3a30003016094191...@10.4.4.251> >;tag=Hr4mHDUeBSNyH Call-ID: 1-6...@10.4.4.252 CSeq: 1 INVITE Contact: <sip:12345616094191...@pgw01.ot.hr:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809: ------------------------------------------------------------------------ ACK sip:30003016094191...@10.4.4.251:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport To: "30003016094191500"<sip:30003016094191...@10.4.4.251<sip%3a30003016094191...@10.4.4.251> > From: "22222238515000403"<sip:22222238515000...@10.4.4.251<sip%3a22222238515000...@10.4.4.251> >;tag=1 Call-ID: 1-6...@10.4.4.252 CSeq: 1 ACK Contact: sip:s...@10.4.4.252:5060 Max-Forwards: 70 Subject: Performance Test Content-Length: 0 What m'i missing ? > Your ACK message must not be valid (dialog matching or something else) > so every 1 call will generate 30 retries that are queued up in the sip > stack. > > at 100cps you will be generating this problem 100 times per second and > queue up countless unfinished dialogs thus > eating up the cpu. > > > > > > > On Mon, Aug 24, 2009 at 12:19 PM, Tihomir Culjaga <tculj...@gmail.com>wrote: > >> Hello, >> >> I've been with freeswittch for a while now.. and i can say it is worth >> developing it. >> >> anyhow i got into a strange issue... I'm tryng to see what load FS on my >> server can take. The Call flow is like this: >> >> SIPp FS >> >> INVITE --------> >> <------- 100 Trying >> <------- 302 Moved Temporary >> ACK ---------> >> >> >> >> I use a dummy dialplan for that. All custom functions i've build are >> disabled and i'm not using it here. Also custom modules are not loaded as >> well. >> >> >> <extension name="ServiceLookup"> >> <condition field="destination_number" expression="(^300030)(.*)"> >> <!--action application="lookup_service_destination" data="in >> ${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, i >> n $1, in pgw01.ot.hr:5060, out red_contact, out authResult"/--> >> <action application="log" data="INFO ######################## >> ServiceLookup ########################\n"/> >> <action application="log" data="INFO ######################## >> contact = '${red_contact}' ##############\n"/> >> <action application="log" data="INFO ######################## >> CallerNum = '${caller_id_number:6:16}' ##########\n"/> >> <action application="log" data="INFO ######################## >> RADIUS auth = '${authResult}' ##########\n"/> >> <action application="execute_extension" data="doRedirect XML >> public"/> >> </condition> >> </extension> >> >> >> <extension name="doRedirect"> >> <condition field="destination_number" expression="^doRedirect$"/> >> <condition field="${authResult}" expression="^0$|^60$"> >> <action application="log" data="INFO ######################## >> RADIUS auth OK!!!' ##########\n"/> >> <!--action application="redirect" data="sip:${red_contact}"/--> >> <!--action application="answer"/--> >> <action application="redirect" data=" >> sip:12345616094191...@pgw01.ot.hr:5060"/> >> <!--anti-action application="answer"/--> >> <!--anti-action application="sleep" data="2000"/--> >> <action application="hangup" data="USER_BUSY"/> >> <anti-action application="redirect" data=" >> sip:12345616094191...@pgw01.ot.hr:5060"/> >> <anti-action application="log" data="INFO >> ######################## RADIUS auth NOK!! ##########\n"/> >> <!--anti-action application="respond" data="403 Forbidden"/--> >> <anti-action application="hangup" data="USER_BUSY"/> >> </condition> >> </extension> >> >> >> When i place a call from x-lite everything works fine ... x-lite sends an >> invite, gets SIP 302 and ACKs it correctly... FS is happy. >> >> When i place a call from SIPp i have the same scenario except FS seems not >> understand ACK message from SIPp and re-sends SIP 302 multiple times untill >> it gives up. >> >> >> I beleive this is due to 302 resend issue but; when i load FS with 100 >> CPS, i can see high CPU usage (just one thread taking most load... the rest >> does almost nothing) on FS. Also, starting from 40 CPS there is a big delay >> in receiving SIP 302 messages meaning i've sent 6000 calls and so far only >> for half of them got 302 response. >> >> >> Does anybody have a clue ? >> >> >> >> >> >> Here is a trace taken on FS for calls originated from SIPp (sipp -sn uac >> 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 100 >> -trace_msg -inf test.txt -m 1 -l 4000): >> >> freeswi...@l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at >> 16:44:26.527236: >> >> ------------------------------------------------------------------------ >> INVITE >> sip:30003016094191...@10.4.4.251<sip%3a30003016094191...@10.4.4.251>SIP/2.0 >> Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport >> Max-Forwards: 70 >> Contact: >> <sip:22222238515000...@10.4.4.252<sip%3a22222238515000...@10.4.4.252> >> > >> To: >> "30003016094191500"<sip:30003016094191...@10.4.4.251<sip%3a30003016094191...@10.4.4.251> >> > >> From: >> "22222238515000403"<sip:22222238515000...@10.4.4.251<sip%3a22222238515000...@10.4.4.251> >> >;tag=1 >> Call-ID: 1-6...@10.4.4.252 >> CSeq: 1 INVITE >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 131 >> >> v=0 >> o=user1 53655765 2353687637 IN IP4 10.4.4.252 >> s=- >> c=IN IP4 10.4.4.252 >> t=0 0 >> m=audio 6000 RTP/AVP 0 >> a=rtpmap:0 PCMU/8000 >> >> ------------------------------------------------------------------------ >> send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060 >> From: >> "22222238515000403"<sip:22222238515000...@10.4.4.251<sip%3a22222238515000...@10.4.4.251> >> >;tag=1 >> To: >> "30003016094191500"<sip:30003016094191...@10.4.4.251<sip%3a30003016094191...@10.4.4.251> >> > >> Call-ID: 1-6...@10.4.4.252 >> CSeq: 1 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582: >> >> ------------------------------------------------------------------------ >> SIP/2.0 302 Moved Temporarily >> Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060 >> From: >> "22222238515000403"<sip:22222238515000...@10.4.4.251<sip%3a22222238515000...@10.4.4.251> >> >;tag=1 >> To: "30003016094191500" >> <sip:30003016094191...@10.4.4.251<sip%3a30003016094191...@10.4.4.251> >> >;tag=Hr4mHDUeBSNyH >> Call-ID: 1-6...@10.4.4.252 >> CSeq: 1 INVITE >> Contact: <sip:12345616094191...@pgw01.ot.hr:5060> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, presence, dialog, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809: >> >> ------------------------------------------------------------------------ >> ACK sip:30003016094191...@10.4.4.251:5060 SIP/2.0 >> Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport >> To: >> "30003016094191500"<sip:30003016094191...@10.4.4.251<sip%3a30003016094191...@10.4.4.251> >> > >> From: >> "22222238515000403"<sip:22222238515000...@10.4.4.251<sip%3a22222238515000...@10.4.4.251> >> >;tag=1 >> Call-ID: 1-6...@10.4.4.252 >> CSeq: 1 ACK >> Contact: sip:s...@10.4.4.252:5060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:27.037070: >> >> ------------------------------------------------------------------------ >> SIP/2.0 302 Moved Temporarily >> Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060 >> From: >> "22222238515000403"<sip:22222238515000...@10.4.4.251<sip%3a22222238515000...@10.4.4.251> >> >;tag=1 >> To: "30003016094191500" >> <sip:30003016094191...@10.4.4.251<sip%3a30003016094191...@10.4.4.251> >> >;tag=Hr4mHDUeBSNyH >> Call-ID: 1-6...@10.4.4.252 >> CSeq: 1 INVITE >> Contact: <sip:12345616094191...@pgw01.ot.hr:5060> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, presence, dialog, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:28.037063: >> >> ------------------------------------------------------------------------ >> SIP/2.0 302 Moved Temporarily >> Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060 >> From: >> "22222238515000403"<sip:22222238515000...@10.4.4.251<sip%3a22222238515000...@10.4.4.251> >> >;tag=1 >> To: "30003016094191500" >> <sip:30003016094191...@10.4.4.251<sip%3a30003016094191...@10.4.4.251> >> >;tag=Hr4mHDUeBSNyH >> Call-ID: 1-6...@10.4.4.252 >> CSeq: 1 INVITE >> Contact: <sip:12345616094191...@pgw01.ot.hr:5060> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, presence, dialog, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer >> Content-Length: 0 >> >> >> Tihomir. >> >> >> >> >> >> _______________________________________________ >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_miness...@hotmail.com <msn%3aanthony_miness...@hotmail.com> > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com<paypal%3aanthony.miness...@gmail.com> > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:8...@conference.freeswitch.org <sip%3a...@conference.freeswitch.org> > iax:gu...@conference.freeswitch.org/888 > googletalk:conf+...@conference.freeswitch.org<googletalk%3aconf%2b...@conference.freeswitch.org> > pstn:213-799-1400 > > > ---------- Forwarded message ---------- > From: "Raffaele P. Guidi" <raffaele.p.gu...@gmail.com> > To: freeswitch-users@lists.freeswitch.org > Date: Mon, 24 Aug 2009 20:24:28 +0200 > Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great > but I have a little problem > Actually I did that and it worked fine. I had the problem the SECOND time I > run FS and freepbx. And (@Brian) mod_sofia was loaded but sip_profiles were > not > > On Sun, Aug 16, 2009 at 16:04, Carlos Talbot <carlos.tal...@gmail.com>wrote: > >> When you configure FreePBX for the first time it wipes out the >> sip_profiles directory. If you follow the FreePBX shortcut on your desktop >> it'll mention this on the last screen of the configuration. You might see >> something such as the following below. If you plan to use FreePBX you'll >> need to define trunk groups, trunks, etc in order to have the sip_profiles >> directory populated. >> regards, >> >> Carlos >> >> >> Incompatible Configuration WARNING: THE FOLLOWING FILES WILL BE DELETED! >> >> - D:/FreeSWITCH/conf/sip_profiles/external.xml >> - D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml >> - D:/FreeSWITCH/conf/sip_profiles/internal.xml >> >> >> On Sun, Aug 16, 2009 at 4:43 AM, Raffaele P. Guidi < >> raffaele.p.gu...@gmail.com> wrote: >> >>> I had the sweet surprise to find the installer packaged with FreePBX... >>> really great! Why it has not been advertised as it deserves? It worked like >>> a breeze once launched, with the automatic configuration and all of that., >>> Only thing: once stopped I cannot get it to load sofia profiles anymore - >>> issueing sofia status doesn't show anything. I had to copy internal.xml and >>> default.xml from a previous installation and everything got to work again - >>> but no FreePBX anymore :( I'm sure I'm missing something important. >>> Can you give me a hint? Should sofia profiles be served by curl or >>> something? >>> >>> Thanks, >>> Raffaele >>> >>> _______________________________________________ >>> 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 >>> >>> >> >> _______________________________________________ >> 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 >> >> > > _______________________________________________ > 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 > >
_______________________________________________ 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