[Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message
David Knell
dave at 3c.co.uk
Mon Aug 24 13:41:23 PDT 2009
Hi Tihomir -
I'm no SIP guru, but the things which look suspicious about the ACK to
me are:
- Via header - different branch
- Contact header - differs from INVITE
--Dave
> 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 ?
>
>
> freeswitch at l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at
> 16:44:26.527236:
> ------------------------------
> ------------------------------------------
> INVITE sip:30003016094191500 at 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:22222238515000403 at 10.4.4.252>
> To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
> From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
> Call-ID: 1-6962 at 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:22222238515000403 at 10.4.4.251>;tag=1
> To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
> Call-ID: 1-6962 at 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:22222238515000403 at 10.4.4.251>;tag=1
> To: "30003016094191500"
> <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
> Call-ID: 1-6962 at 10.4.4.252
> CSeq: 1 INVITE
> Contact: <sip:12345616094191500 at 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:30003016094191500 at 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:30003016094191500 at 10.4.4.251>
> From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
> Call-ID: 1-6962 at 10.4.4.252
> CSeq: 1 ACK
> Contact: sip:sipp at 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
> <tculjaga at 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:12345616094191500 at 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:12345616094191500 at 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):
>
> freeswitch at l01sipindir1> recv 573 bytes from
> udp/[10.4.4.252]:5060 at 16:44:26.527236:
>
> ------------------------------------------------------------------------
> INVITE sip:30003016094191500 at 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:22222238515000403 at 10.4.4.252>
> To:
> "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
> From:
> "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
> Call-ID: 1-6962 at 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:22222238515000403 at 10.4.4.251>;tag=1
> To:
> "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
> Call-ID: 1-6962 at 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:22222238515000403 at 10.4.4.251>;tag=1
> To: "30003016094191500"
> <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
> Call-ID: 1-6962 at 10.4.4.252
> CSeq: 1 INVITE
> Contact: <sip:12345616094191500 at 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:30003016094191500 at 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:30003016094191500 at 10.4.4.251>
> From:
> "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
> Call-ID: 1-6962 at 10.4.4.252
> CSeq: 1 ACK
> Contact: sip:sipp at 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:22222238515000403 at 10.4.4.251>;tag=1
> To: "30003016094191500"
> <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
> Call-ID: 1-6962 at 10.4.4.252
> CSeq: 1 INVITE
> Contact: <sip:12345616094191500 at 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:22222238515000403 at 10.4.4.251>;tag=1
> To: "30003016094191500"
> <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
> Call-ID: 1-6962 at 10.4.4.252
> CSeq: 1 INVITE
> Contact: <sip:12345616094191500 at 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 at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
>
> ---------- Forwarded message ----------
> From: "Raffaele P. Guidi" <raffaele.p.guidi at gmail.com>
> To: freeswitch-users at 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.talbot at 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.guidi at 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
>
>
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>
>
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>
>
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--
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk
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