[Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message
Tihomir Culjaga
tculjaga at gmail.com
Mon Aug 24 12:03:01 PDT 2009
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%3A30003016094191500 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<sip%3A22222238515000403 at 10.4.4.252>
>
To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 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<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 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<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
To: "30003016094191500"
<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 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<sip%3A30003016094191500 at 10.4.4.251>
>
From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 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%3A30003016094191500 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<sip%3A22222238515000403 at 10.4.4.252>
>> >
>> To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>> >
>> From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 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<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>> To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 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<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>> To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 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<sip%3A30003016094191500 at 10.4.4.251>
>> >
>> From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 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<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>> To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 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<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>> To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 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
>> 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_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 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
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>
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