> Hi,
> Out of curiosity, I tried this (-trace_calldebug) flag and sipp
> complaint that no such flag available.
>
> # sipp -v
> SIPp v3.1-TLS-PCAP, version svn9910, built May 2 2008, 15:31:41.
>
> Am I using the latest version?
> Thanks and Regards,
> Manish
>
I´m using "sipp.2008-09-25". You can find the latest version in
http://sipp.sourceforge.net/snapshots/
>
>
> ZiLi0n wrote:
> > Mr. Charles, I have seen "hash 0" in calldebug when SIPp sends the message
> > "BYE" :
> >
> > 2008-10-10 08:12:14:785 1223619134.785339 Set Retransmission Hash: 0
> > (recv index 3, send index 9)
> > 2008-10-10 08:12:14:785 1223619134.785348 Processing message 10 of
> > type 3 for call [EMAIL PROTECTED] at 183076.
> > 2008-10-10 08:12:15:295 1223619135.295335 Processing message 10 of
> > type 3 for call [EMAIL PROTECTED] at 183586.
> > 2008-10-10 08:12:15:295 1223619135.295344 Retransmisison required (1
> > retransmissions, max 9)
> > 2008-10-10 08:12:15:295 1223619135.295362 Sending UDP message for
> > call [EMAIL PROTECTED] (index 9, hash 502009980):
> >
> > Thanks a lot!
> >
> >> To: [EMAIL PROTECTED]
> >> From: [EMAIL PROTECTED]
> >> Date: Thu, 9 Oct 2008 11:42:54 -0400
> >> CC: [email protected]
> >> Subject: Re: [Sipp-users] Why doesn´t SIPp re-send the ACK when it receive
> >> a lot of messages 200 OK?
> >>
> >> Ths is a -trace_msg trace not a -trace_calldebug trace.
> >>
> >> Charles
> >>
> >>
> >>
> >>
> >> ZiLi0n <[EMAIL PROTECTED]>
> >> 10/09/2008 11:27 AM
> >>
> >> To
> >> <[email protected]>
> >> cc
> >>
> >> Subject
> >> [Sipp-users] Re: Why doesn´t SIPp re-send the ACK when it receive a lot
> >> of messages 200 OK?
> >>
> >>
> >>
> >>
> >>
> >>
> >> Mr Charles. Thank. Here is the traces
> >>
> >> In this traces, SIPp don´t resends the ACK even when Asterisk resends the
> >> message '200 OK':
> >>
> >> ----------------------------------------------- 2008-10-08
> >> 13:11:08:995.378
> >> UDP message received [758] bytes :
> >>
> >> SIP/2.0 200 OK^M
> >> Via: SIP/2.0/UDP
> >> 192.168.1.179:5060;branch=z9hG4bK-2882-230-0;received=192.168.1.179^M
> >> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=2882SIPpTag00230^M
> >> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as7fc476b0^M
> >> Call-ID: [EMAIL PROTECTED]
> >> CSeq: 1 INVITE^M
> >> User-Agent: Asterisk PBX 1.6.0^M
> >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M
> >> Supported: replaces, timer^M
> >> Contact: <sip:[EMAIL PROTECTED]>^M
> >> Content-Type: application/sdp^M
> >> Content-Length: 261^M
> >> ^M
> >> v=0^M
> >> o=root 377264768 377264768 IN IP4 192.168.1.246^M
> >> s=Asterisk PBX 1.6.0^M
> >> c=IN IP4 192.168.1.246^M
> >> t=0 0^M
> >> m=audio 18590 RTP/AVP 8 101^M
> >> a=rtpmap:8 PCMA/8000^M
> >> a=rtpmap:101 telephone-event/8000^M
> >> a=fmtp:101 0-16^M
> >> a=silenceSupp:off - - - -^M
> >> a=ptime:20^M
> >> a=sendrecv^M
> >>
> >> ----------------------------------------------- 2008-10-08
> >> 13:11:08:995.480
> >> UDP message sent (383 bytes):
> >>
> >> ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0^M
> >> Via: SIP/2.0/UDP 192.168.1.179:5060;branch=z9hG4bK-2882-230-4^M
> >> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=2882SIPpTag00230^M
> >> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as7fc476b0^M
> >> Call-ID: [EMAIL PROTECTED]
> >> CSeq: 1 ACK^M
> >> Contact: sip:[EMAIL PROTECTED]:5060^M
> >> Max-Forwards: 70^M
> >> Subject: Performance Test^M
> >> Content-Length: 0^M
> >> ^M
> >> ----------------------------------------------- 2008-10-08
> >> 13:11:09:993.684
> >> UDP message received [758] bytes :
> >>
> >> SIP/2.0 200 OK^M
> >> Via: SIP/2.0/UDP
> >> 192.168.1.179:5060;branch=z9hG4bK-2882-230-0;received=192.168.1.179^M
> >> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=2882SIPpTag00230^M
> >> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as7fc476b0^M
> >> Call-ID: [EMAIL PROTECTED]
> >> CSeq: 1 INVITE^M
> >> User-Agent: Asterisk PBX 1.6.0^M
> >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M
> >> Supported: replaces, timer^M
> >> Contact: <sip:[EMAIL PROTECTED]>^M
> >> Content-Type: application/sdp^M
> >> Content-Length: 261^M
> >> ^M
> >> v=0^M
> >> o=root 377264768 377264768 IN IP4 192.168.1.246^M
> >> s=Asterisk PBX 1.6.0^M
> >> c=IN IP4 192.168.1.246^M
> >> t=0 0^M
> >> m=audio 18590 RTP/AVP 8 101^M
> >> a=rtpmap:8 PCMA/8000^M
> >> a=rtpmap:101 telephone-event/8000^M
> >> a=fmtp:101 0-16^M
> >> a=silenceSupp:off - - - -^M
> >> a=ptime:20^M
> >> a=sendrecv^M
> >>
> >> ----------------------------------------------- 2008-10-08
> >> 13:11:10:994.064
> >> UDP message received [758] bytes :
> >>
> >> SIP/2.0 200 OK^M
> >> Via: SIP/2.0/UDP
> >> 192.168.1.179:5060;branch=z9hG4bK-2882-230-0;received=192.168.1.179^M
> >> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=2882SIPpTag00230^M
> >> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as7fc476b0^M
> >> Call-ID: [EMAIL PROTECTED]
> >> CSeq: 1 INVITE^M
> >> User-Agent: Asterisk PBX 1.6.0^M
> >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M
> >> Supported: replaces, timer^M
> >> Contact: <sip:[EMAIL PROTECTED]>^M
> >> Content-Type: application/sdp^M
> >> Content-Length: 261^M
> >> ^M
> >> v=0^M
> >> o=root 377264768 377264768 IN IP4 192.168.1.246^M
> >> s=Asterisk PBX 1.6.0^M
> >> c=IN IP4 192.168.1.246^M
> >> t=0 0^M
> >> m=audio 18590 RTP/AVP 8 101^M
> >> a=rtpmap:8 PCMA/8000^M
> >> a=rtpmap:101 telephone-event/8000^M
> >>
> >>
> >> Other problem is SIPp retrans the message BYE even when Asterisk has sent
> >> the '200 OK':
> >>
> >> U +0.026130 192.168.1.136:5060 -> 192.168.1.246:5060
> >> BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
> >> Via: SIP/2.0/UDP 192.168.1.136:5060;branch=z9hG4bK-12878-227-9
> >> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=12878SIPpTag00227
> >> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as7e62485f
> >> Call-ID: [EMAIL PROTECTED]
> >> CSeq: 2 BYE
> >> Contact: sip:[EMAIL PROTECTED]:5060
> >> Max-Forwards: 70
> >> Subject: Performance Test
> >> Content-Length: 0
> >>
> >>
> >> #
> >> U +0.000146 192.168.1.246:5060 -> 192.168.1.136:5060
> >> SIP/2.0 200 OK
> >> Via: SIP/2.0/UDP
> >> 192.168.1.136:5060;branch=z9hG4bK-12878-227-9;received=192.168.1.136
> >> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=12878SIPpTag00227
> >> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as7e62485f
> >> Call-ID: [EMAIL PROTECTED]
> >> CSeq: 2 BYE
> >> User-Agent: Asterisk PBX 1.6.0
> >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> Supported: replaces, timer
> >> Contact: <sip:[EMAIL PROTECTED]>
> >> Content-Length: 0
> >>
> >> U +0.152849 192.168.1.136:5060 -> 192.168.1.246:5060
> >> BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
> >> Via: SIP/2.0/UDP 192.168.1.136:5060;branch=z9hG4bK-12878-227-9
> >> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=12878SIPpTag00227
> >> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as7e62485f
> >> Call-ID: [EMAIL PROTECTED]
> >> CSeq: 2 BYE
> >> Contact: sip:[EMAIL PROTECTED]:5060
> >> Max-Forwards: 70
> >> Subject: Performance Test
> >> Content-Length: 0
> >>
> >>
> >> #
> >> U +0.000125 192.168.1.246:5060 -> 192.168.1.136:5060
> >> SIP/2.0 481 Call leg/transaction does not exist
> >> Via: SIP/2.0/UDP
> >> 192.168.1.136:5060;branch=z9hG4bK-12878-227-9;received=192.168.1.136
> >> From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=12878SIPpTag00227
> >> To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as7e62485f
> >> Call-ID: [EMAIL PROTECTED]
> >> CSeq: 2 BYE
> >> User-Agent: Asterisk PBX 1.6.0
> >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> >> Supported: replaces, timer
> >> Content-Length: 0
> >>
> >> I think that SIPp CLIENT hasn´t got time for receive the response from
> >> Asterisk. this errors only occurrs when the load is heavy.
> >>
> >> Some solution?
> >>
> >> PD: For this test I use sipp.2008-09-25.tar
> >>
> >> Thanks.
> >>
> >>> To: [EMAIL PROTECTED]
> >>> From: [EMAIL PROTECTED]
> >>> Date: Tue, 7 Oct 2008 19:36:24 -0400
> >>> CC: [email protected]
> >>> Subject: Re: [Sipp-users] Why doesn´t SIPp re-send the ACK when it
> >> receive a lot of messages 200 OK?
> >>> You should include the output of -trace_calldebug in your message.
> >>>
> >>> Charles
> >>>
> >>> ZiLi0n <[EMAIL PROTECTED]> wrote on 10/07/2008 04:33:10 PM:
> >>>
> >>>>> SIPp should retransmit the ACK if the retransmitted 200 is
> >> identical.
> >>> You
> >>>>> should enable -trace_msg and -trace_calldebug and see what the logs
> >>> tell
> >>>>> you.
> >>>> The retrasnmitted meesages 200 OK are identicals.
> >>>>
> >>>> I´m using SIPp with Asterisk.Asterisk show this errors:
> >>>> Sep 12 18:52:36 WARNING [4620]: chan_sip.c:1835 retrans_pkt: Maximum
> >>>> retries exceeded on transmission 778f89593967725f0abe40eb1752504c (at)
> >>>> 10.10.206.53 for seqno 1620 (Critical Response)
> >>>>
> >>>> Sep 12 18:52:36 WARNING [4620]: chan_sip.c:1835 retrans_pkt: Hanging
> >>>> up call 778f89593967725f0abe40eb1752504c (at) 10.10.206.53 no reply to
> >>>> our critical packet.
> >>>>
> >>>> A solution? Thanks
> >>>>
> >>>>> Charles
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>> ZiLi0n <[EMAIL PROTECTED]>
> >>>>> 10/05/2008 02:26 PM
> >>>>>
> >>>>> To
> >>>>> <[email protected]>
> >>>>> cc
> >>>>>
> >>>>> Subject
> >>>>> [Sipp-users] Why doesn´t SIPp re-send the ACK when it receive a lot
> >> of
> >>>>> messages 200 OK?
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>> I am testing my Asterisk Server. This is my configuration:
> >>>>>
> >>>>> SIPpCLIENT --- Asterisk Server --- SIPpSERVER
> >>>>>
> >>>>> If the simultaneous calls is less than 300 the test is OK, but when
> >>> the
> >>>>> simultaneous calls is approximately 350 calls, the calls hang up:
> >>>>>
> >>>>> Asterisk sends to SIPpCLIENT the message '200 OK'. SIPpCLIENT sends
> >> to
> >>>>> Asterisk the message 'ACK'.
> >>>>> Asterisk re-sends to SIPpCLIENT the message '200 OK'.
> >>>>> SIPpCLIENT doesn´t re-send the message 'ACK' to Asterisk.
> >>>>>
> >>>>> I think that message 'ACK' is lost... but the cpu load Asterisk
> >> Server
> >>> is
> >>>>> approximately 50% and the net status is perfect.
> >>>>>
> >>>>> Why does not SIPpCLIENT re-send the ACK to Asterisk? Asterisk needs
> >>> the
> >>>>> ACK for complete the call
> >>>>>
> >>>>> Thank´s
> >>>>>
> >>>>> La cartera, las gafas. ¿te falta algo? Ahora llévate Messenger en tu
> >>> móvil
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> >>>>> world
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> >>>>> _______________________________________________
> >>>>> Sipp-users mailing list
> >>>>> [email protected]
> >>>>> https://lists.sourceforge.net/lists/listinfo/sipp-users
> >>>>>
> >>>>>
> >>>>>
> >>>>>
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> >>>>> [email protected]
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> >>>> _______________________________________________
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> >>>> [email protected]
> >>>> https://lists.sourceforge.net/lists/listinfo/sipp-users
> >>>
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> >>> _______________________________________________
> >>> Sipp-users mailing list
> >>> [email protected]
> >>> https://lists.sourceforge.net/lists/listinfo/sipp-users
> >> Juega a las preguntas de Live Quiz con tus contactos de Messenger ¡Empieza
> >> ahora!
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> >> _______________________________________________
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> >>
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> >> _______________________________________________
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> >> [email protected]
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> >
> > _________________________________________________________________
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