Hi Anthony,<br><br>I'm aware it is generating 30 retries per a call and this is killing me ...<br><br>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.<br>
<br>pls can you help ?<br><br><br>freeswitch@l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at 16:44:26.527236:<br> ------------------------------<div class="gmail_quote">------------------------------------------<br>
INVITE <a href="mailto:sip%3A30003016094191500@10.4.4.251" target="_blank">sip:30003016094191500@10.4.4.251</a> SIP/2.0<br>
Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport<br> Max-Forwards: 70<br> Contact: <<a href="mailto:sip%3A22222238515000403@10.4.4.252" target="_blank">sip:22222238515000403@10.4.4.252</a>><br> To: "30003016094191500"<<a href="mailto:sip%3A30003016094191500@10.4.4.251" target="_blank">sip:30003016094191500@10.4.4.251</a>><br>
From: "22222238515000403"<<a href="mailto:sip%3A22222238515000403@10.4.4.251" target="_blank">sip:22222238515000403@10.4.4.251</a>>;tag=1<br> Call-ID: <a href="mailto:1-6962@10.4.4.252" target="_blank">1-6962@10.4.4.252</a><br>
CSeq: 1 INVITE<br>
Max-Forwards: 70<br> Subject: Performance Test<br> Content-Type: application/sdp<br> Content-Length: 131<br> <br> v=0<br> o=user1 53655765 2353687637 IN IP4 10.4.4.252<br> s=-<br> c=IN IP4 10.4.4.252<br>
t=0 0<br> m=audio 6000 RTP/AVP 0<br> a=rtpmap:0 PCMU/8000<br> ------------------------------------------------------------------------<br>send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566:<br> ------------------------------------------------------------------------<br>
SIP/2.0 100 Trying<br> Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060<br> From: "22222238515000403"<<a href="mailto:sip%3A22222238515000403@10.4.4.251" target="_blank">sip:22222238515000403@10.4.4.251</a>>;tag=1<br>
To: "30003016094191500"<<a href="mailto:sip%3A30003016094191500@10.4.4.251" target="_blank">sip:30003016094191500@10.4.4.251</a>><br> Call-ID: <a href="mailto:1-6962@10.4.4.252" target="_blank">1-6962@10.4.4.252</a><br>
CSeq: 1 INVITE<br>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported<br> Content-Length: 0<br> <br> ------------------------------------------------------------------------<br>send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582:<br>
------------------------------------------------------------------------<br> SIP/2.0 302 Moved Temporarily<br> Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060<br> From: "22222238515000403"<<a href="mailto:sip%3A22222238515000403@10.4.4.251" target="_blank">sip:22222238515000403@10.4.4.251</a>>;tag=1<br>
To: "30003016094191500" <<a href="mailto:sip%3A30003016094191500@10.4.4.251" target="_blank">sip:30003016094191500@10.4.4.251</a>>;tag=Hr4mHDUeBSNyH<br> Call-ID: <a href="mailto:1-6962@10.4.4.252" target="_blank">1-6962@10.4.4.252</a><br>
CSeq: 1 INVITE<br> Contact: <<a href="http://sip:12345616094191500@pgw01.ot.hr:5060/" target="_blank">sip:12345616094191500@pgw01.ot.hr:5060</a>><br> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported<br> Accept: application/sdp<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH<br> Supported: timer, precondition, path, replaces<br> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer<br>
Content-Length: 0<br><br><br> ------------------------------------------------------------------------<br><div class="gmail_quote">recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:<br> ------------------------------------------------------------------------<br>
ACK <a href="http://sip:30003016094191500@10.4.4.251:5060/" target="_blank">sip:30003016094191500@10.4.4.251:5060</a> SIP/2.0<br> Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport<br> To: "30003016094191500"<<a href="mailto:sip%3A30003016094191500@10.4.4.251" target="_blank">sip:30003016094191500@10.4.4.251</a>><br>
From: "22222238515000403"<<a href="mailto:sip%3A22222238515000403@10.4.4.251" target="_blank">sip:22222238515000403@10.4.4.251</a>>;tag=1<br> Call-ID: <a href="mailto:1-6962@10.4.4.252" target="_blank">1-6962@10.4.4.252</a><br>
CSeq: 1 ACK<br>
Contact: <a href="http://sip:sipp@10.4.4.252:5060/" target="_blank">sip:sipp@10.4.4.252:5060</a><br> Max-Forwards: 70<br> Subject: Performance Test<br> Content-Length: 0<br><br></div><br><br><br>What m'i missing ?<br>
</div><br><br><div class="gmail_quote"><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>Your ACK message must not be valid (dialog matching or something else)<br>
so every 1 call will generate 30 retries that are queued up in the sip stack.<br><br>at 100cps you will be generating this problem 100 times per second and queue up countless unfinished dialogs thus<br>
eating up the cpu.<br><br><br><br><br><br><br><div class="gmail_quote">On Mon, Aug 24, 2009 at 12:19 PM, Tihomir Culjaga <span dir="ltr"><<a href="mailto:tculjaga@gmail.com" target="_blank">tculjaga@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div class="gmail_quote">Hello,<br><br>I've been with freeswittch for a while now.. and i can say it is worth developing it.<br>
<br>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:<br>
<br>SIPp FS<br><br>INVITE --------> <br> <------- 100 Trying<br> <------- 302 Moved Temporary<br>ACK ---------><br><br><br><br>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.<br>
<br><br> <extension name="ServiceLookup"><br> <condition field="destination_number" expression="(^300030)(.*)"><br> <!--action application="lookup_service_destination" data="in ${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, i<br>
n $1, in <a href="http://pgw01.ot.hr:5060" target="_blank">pgw01.ot.hr:5060</a>, out red_contact, out authResult"/--><br> <action application="log" data="INFO ######################## ServiceLookup ########################\n"/><br>
<action application="log" data="INFO ######################## contact = '${red_contact}' ##############\n"/><br> <action application="log" data="INFO ######################## CallerNum = '${caller_id_number:6:16}' ##########\n"/><br>
<action application="log" data="INFO ######################## RADIUS auth = '${authResult}' ##########\n"/><br> <action application="execute_extension" data="doRedirect XML public"/><br>
</condition><br> </extension><br><br><br> <extension name="doRedirect"><br> <condition field="destination_number" expression="^doRedirect$"/><br> <condition field="${authResult}" expression="^0$|^60$"><br>
<action application="log" data="INFO ######################## RADIUS auth OK!!!' ##########\n"/><br> <!--action application="redirect" data="sip:${red_contact}"/--><br>
<!--action application="answer"/--><br> <action application="redirect" data="<a href="http://sip:12345616094191500@pgw01.ot.hr:5060" target="_blank">sip:12345616094191500@pgw01.ot.hr:5060</a>"/><br>
<!--anti-action application="answer"/--><br> <!--anti-action application="sleep" data="2000"/--><br> <action application="hangup" data="USER_BUSY"/><br>
<anti-action application="redirect" data="<a href="http://sip:12345616094191500@pgw01.ot.hr:5060" target="_blank">sip:12345616094191500@pgw01.ot.hr:5060</a>"/><br> <anti-action application="log" data="INFO ######################## RADIUS auth NOK!! ##########\n"/><br>
<!--anti-action application="respond" data="403 Forbidden"/--><br> <anti-action application="hangup" data="USER_BUSY"/><br> </condition><br>
</extension><br>
<br><br>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.<br><br>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.<br>
<br><br>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.<br><br><br>Does anybody have a clue ?<br><br><br><br><br><br>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):<br>
<br>freeswitch@l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at 16:44:26.527236:<br> ------------------------------------------------------------------------<br> INVITE <a href="mailto:sip%3A30003016094191500@10.4.4.251" target="_blank">sip:30003016094191500@10.4.4.251</a> SIP/2.0<br>
Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport<br> Max-Forwards: 70<br> Contact: <<a href="mailto:sip%3A22222238515000403@10.4.4.252" target="_blank">sip:22222238515000403@10.4.4.252</a>><br> To: "30003016094191500"<<a href="mailto:sip%3A30003016094191500@10.4.4.251" target="_blank">sip:30003016094191500@10.4.4.251</a>><br>
From: "22222238515000403"<<a href="mailto:sip%3A22222238515000403@10.4.4.251" target="_blank">sip:22222238515000403@10.4.4.251</a>>;tag=1<br> Call-ID: <a href="mailto:1-6962@10.4.4.252" target="_blank">1-6962@10.4.4.252</a><br>
CSeq: 1 INVITE<br>
Max-Forwards: 70<br> Subject: Performance Test<br> Content-Type: application/sdp<br> Content-Length: 131<br> <br> v=0<br> o=user1 53655765 2353687637 IN IP4 10.4.4.252<br> s=-<br> c=IN IP4 10.4.4.252<br>
t=0 0<br> m=audio 6000 RTP/AVP 0<br> a=rtpmap:0 PCMU/8000<br> ------------------------------------------------------------------------<br>send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566:<br> ------------------------------------------------------------------------<br>
SIP/2.0 100 Trying<br> Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060<br> From: "22222238515000403"<<a href="mailto:sip%3A22222238515000403@10.4.4.251" target="_blank">sip:22222238515000403@10.4.4.251</a>>;tag=1<br>
To: "30003016094191500"<<a href="mailto:sip%3A30003016094191500@10.4.4.251" target="_blank">sip:30003016094191500@10.4.4.251</a>><br> Call-ID: <a href="mailto:1-6962@10.4.4.252" target="_blank">1-6962@10.4.4.252</a><br>
CSeq: 1 INVITE<br>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported<br> Content-Length: 0<br> <br> ------------------------------------------------------------------------<br>send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582:<br>
------------------------------------------------------------------------<br> SIP/2.0 302 Moved Temporarily<br> Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060<br> From: "22222238515000403"<<a href="mailto:sip%3A22222238515000403@10.4.4.251" target="_blank">sip:22222238515000403@10.4.4.251</a>>;tag=1<br>
To: "30003016094191500" <<a href="mailto:sip%3A30003016094191500@10.4.4.251" target="_blank">sip:30003016094191500@10.4.4.251</a>>;tag=Hr4mHDUeBSNyH<br> Call-ID: <a href="mailto:1-6962@10.4.4.252" target="_blank">1-6962@10.4.4.252</a><br>
CSeq: 1 INVITE<br> Contact: <<a href="http://sip:12345616094191500@pgw01.ot.hr:5060" target="_blank">sip:12345616094191500@pgw01.ot.hr:5060</a>><br> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported<br> Accept: application/sdp<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH<br> Supported: timer, precondition, path, replaces<br> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer<br>
Content-Length: 0<br> <br> ------------------------------------------------------------------------<br>recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:<br> ------------------------------------------------------------------------<br>
ACK <a href="http://sip:30003016094191500@10.4.4.251:5060" target="_blank">sip:30003016094191500@10.4.4.251:5060</a> SIP/2.0<br> Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport<br> To: "30003016094191500"<<a href="mailto:sip%3A30003016094191500@10.4.4.251" target="_blank">sip:30003016094191500@10.4.4.251</a>><br>
From: "22222238515000403"<<a href="mailto:sip%3A22222238515000403@10.4.4.251" target="_blank">sip:22222238515000403@10.4.4.251</a>>;tag=1<br> Call-ID: <a href="mailto:1-6962@10.4.4.252" target="_blank">1-6962@10.4.4.252</a><br>
CSeq: 1 ACK<br>
Contact: <a href="http://sip:sipp@10.4.4.252:5060" target="_blank">sip:sipp@10.4.4.252:5060</a><br> Max-Forwards: 70<br> Subject: Performance Test<br> Content-Length: 0<br> <br> ------------------------------------------------------------------------<br>
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:27.037070:<br> ------------------------------------------------------------------------<br> SIP/2.0 302 Moved Temporarily<br> Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060<br>
From: "22222238515000403"<<a href="mailto:sip%3A22222238515000403@10.4.4.251" target="_blank">sip:22222238515000403@10.4.4.251</a>>;tag=1<br> To: "30003016094191500" <<a href="mailto:sip%3A30003016094191500@10.4.4.251" target="_blank">sip:30003016094191500@10.4.4.251</a>>;tag=Hr4mHDUeBSNyH<br>
Call-ID: <a href="mailto:1-6962@10.4.4.252" target="_blank">1-6962@10.4.4.252</a><br> CSeq: 1 INVITE<br> Contact: <<a href="http://sip:12345616094191500@pgw01.ot.hr:5060" target="_blank">sip:12345616094191500@pgw01.ot.hr:5060</a>><br>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported<br>
Accept: application/sdp<br> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH<br> Supported: timer, precondition, path, replaces<br> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer<br>
Content-Length: 0<br> <br> ------------------------------------------------------------------------<br>send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:28.037063:<br> ------------------------------------------------------------------------<br>
SIP/2.0 302 Moved Temporarily<br> Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060<br> From: "22222238515000403"<<a href="mailto:sip%3A22222238515000403@10.4.4.251" target="_blank">sip:22222238515000403@10.4.4.251</a>>;tag=1<br>
To: "30003016094191500" <<a href="mailto:sip%3A30003016094191500@10.4.4.251" target="_blank">sip:30003016094191500@10.4.4.251</a>>;tag=Hr4mHDUeBSNyH<br> Call-ID: <a href="mailto:1-6962@10.4.4.252" target="_blank">1-6962@10.4.4.252</a><br>
CSeq: 1 INVITE<br> Contact: <<a href="http://sip:12345616094191500@pgw01.ot.hr:5060" target="_blank">sip:12345616094191500@pgw01.ot.hr:5060</a>><br> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported<br> Accept: application/sdp<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH<br> Supported: timer, precondition, path, replaces<br> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer<br>
Content-Length: 0<br><font color="#888888"><br><br>Tihomir.<br><br><br><br>
</font></div><br>
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<br></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
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GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com" target="_blank">PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a><br>
<a href="http://iax:guest@conference.freeswitch.org/888" target="_blank">iax:guest@conference.freeswitch.org/888</a><br>
<a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org" target="_blank">googletalk:conf+888@conference.freeswitch.org</a><br>pstn:213-799-1400<br>
<br><br>---------- Forwarded message ----------<br>From: "Raffaele P. Guidi" <<a href="mailto:raffaele.p.guidi@gmail.com">raffaele.p.guidi@gmail.com</a>><br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
Date: Mon, 24 Aug 2009 20:24:28 +0200<br>Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem<br>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<br>
<br><div class="gmail_quote">On Sun, Aug 16, 2009 at 16:04, Carlos Talbot <span dir="ltr"><<a href="mailto:carlos.talbot@gmail.com" target="_blank">carlos.talbot@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">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.<div>
<br></div><div>regards,</div><div><br></div><div>Carlos</div><div><br></div><div><br></div><div><span style="font-family: -webkit-sans-serif; font-size: 12px;"><table style="margin: 0px; border-collapse: collapse; width: 748px;" cellpadding="0">
<tbody><tr><th colspan="2" style="border-top: 2px solid rgb(198, 12, 48); border-bottom: 1px solid rgb(221, 221, 221); padding: 5px; color: rgb(198, 12, 48); font-size: 14px; font-weight: bold; text-align: left;">
Incompatible Configuration</th></tr><tr><td colspan="2" style="padding: 5px 10px; font-size: 12px; border-bottom-width: 1px; border-bottom-color: rgb(221, 221, 221);">
<span style="text-align: center;"><h3>WARNING: THE FOLLOWING FILES WILL BE DELETED!</h3><br><ul style="text-align: left; font-size: 1em;"><li>D:/FreeSWITCH/conf/sip_profiles/external.xml</li><li>D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml</li>
<li>D:/FreeSWITCH/conf/sip_profiles/internal.xml</li></ul></span></td></tr></tbody></table></span><br><div class="gmail_quote"><div><div></div><div>On Sun, Aug 16, 2009 at 4:43 AM, Raffaele P. Guidi <span dir="ltr"><<a href="mailto:raffaele.p.guidi@gmail.com" target="_blank">raffaele.p.guidi@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div></div><div>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.<div>
<br></div><div>Can you give me a hint? Should sofia profiles be served by curl or something?</div><div><br></div><div>Thanks,</div><div> Raffaele</div>
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