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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