Hi,

no I haven´t used wireshark, yet. Just watched the 
http://sipp.sourceforge.net/doc/uas.xml and 
http://sipp.sourceforge.net/doc/uac_pcap.xml.html for the first time.
I thought those were complete SIP Calls, i.e. REGISTER, INVITE, BYE, ...
But they aren´t :-( Several steps have to be done:

{1. SIPP-Server registers
2. SIPP-Client registers
3. SIPP-Client invites SIPP-Server
3. SIPP Client sends the prerecorded file to the SIPP-Server
4. SIPP-Server echoes back what SIPP-Client formerly sent.
(5. SIPP-Client sends again, SIPP-Server replies again)
6. Session will be ended.
}* x-times

I hope this is everything, do I forget something?
Btw, are there anywhere more premade scenarios, or do I have to make it 
on my own? (Haven´t done this yet before, so that would be quite long...)

Christian



Simon Flannery schrieb:
>
> Are you tracing using ethereal / wireshark? This will help with the 
> capturing the unexpected messages. If you are tracing, what are the 
> unexpected messages? Also, silly question, have you first registered 
> the users and are they successfully registered?
>  
> Simon
>  
> On 8/15/07, *Christian Kraus* <[EMAIL PROTECTED] 
> <mailto:[EMAIL PROTECTED]>> wrote:
>
>     Hi,
>
>     I want to test my Asterisk-Server. Therefore I want to use the
>     included
>     uac_pcap.xml.
>     I´m posting my configs and only changing the IPs. Hopefully anyone
>     could
>     help, as I can´t find something googling for hours/days.
>
>
>     I set up:
>
>     SIPP-Client: 89.111.111.111 <http://89.111.111.111>
>     SIPP-Server: 89.222.222.222 <http://89.222.222.222>
>     Asterisk: 89.555.555.555
>
>
>     on SIPP-Server I run:
>     ./sipp -sn uas -i 89.222.222.222 <http://89.222.222.222>
>     89.222.222.222 <http://89.222.222.222>
>
>     and an SIPP-Client:
>     ./sipp -sf uac_pcap.xml -d 100000 -s 2000 89.555.555.555 -l 1000 -r 10
>     -rp 1000
>
>
>     To the asterisk´s sip.conf I added:
>
>     [sipp]
>     context=test
>     language=en
>     type=friend
>     host=dynamic
>     nat=no
>
>     To the extension.conf I added:
>
>     [test]
>     exten => 2000,1,Dial(SIP/89.222.222.222)
>
>     So every caller being 2000, will be directed to the SIPP-Server...
>
>
>
>     As you can see the SIPP-Client is sending, but it gets "unexpected
>     messages":
>
>     ------------------------------ Scenario Screen -------- [1-9]: Change
>     Screen --
>     Call-rate(length) Port Total-time Total-calls Remote-host
>     10.0(100000 ms)/1.000s 5060 28.01 s 280 89.555.555.555:5060(UDP)
>
>     10 new calls during 1.000 s period 2 ms scheduler resolution
>     127 calls (limit 1000) Peak was 141 calls, after 14 s
>     0 Running, 128 Paused, 10 Woken up
>     0 out-of-call msg (discarded)
>     1 open sockets
>     0 Total RTP pckts sent 0.000 last period RTP rate (kB/s)
>
>     Messages Retrans Timeout Unexpected-Msg
>     INVITE ----------> 280 657 0
>     100 <---------- 0 0 0 153
>     180 <---------- 0 0 0 0
>     200 <---------- E-RTD1 0 0 0 0
>
>     ACK ----------> 0 0
>     [ NOP ]
>     Pause [ 8000ms] 0 0
>     [ NOP ]
>     Pause [ 1000ms] 0 0
>     BYE ----------> 0 0 0
>     200 <---------- 0 0 0 0
>
>     ------ [+|-|*|/]: Adjust rate ---- [q]: Soft exit ---- [p]: Pause
>     traffic -----
>
>
>
>
>     The SIPP-Server is receiving messages and sends answers back:
>
>     ------------------------------ Scenario Screen -------- [1-9]: Change
>     Screen --
>     Port Total-time Total-calls Transport
>     5060 35.02 s 209 UDP
>
>     10 new calls during 1.001 s period 2 ms scheduler resolution
>     164 calls Peak was 165 calls, after 30 s
>     0 Running, 164 Paused, 0 Woken up
>     1 open sockets
>
>     Messages Retrans Timeout Unexpected-Msg
>     ----------> INVITE 209 0 0
>
>     <---------- 180 209 0
>     <---------- 200 209 0 0
>     ----------> ACK E-RTD1 209 0 0
>
>     ----------> BYE 85 0 0
>     <---------- 200 85 0
>     [ 4000ms] Pause 85 0
>     ------------------------------ Sipp Server Mode
>     -------------------------------
>
>
>
>
>     But as you can see on the SIPP-Client´s output, there are no
>     succesfull
>     calls. I logged everything and get the following failure types:
>
>     sipp: The following events occured:
>     2007-08-15 16:38:16:570 1187188696.570840: Aborting call on unexpected
>     message for Call-Id '[EMAIL PROTECTED]
>     <mailto:[EMAIL PROTECTED]>': while$
>     Via: SIP/2.0/UDP
>     127.0.0.1:5060
>     <http://127.0.0.1:5060>;branch=z9hG4bK-5633-96-0;received=89.110.157.78
>     <http://89.110.157.78>
>     From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=96
>     To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as52a75494
>     Call-ID: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>
>     CSeq: 1 INVITE
>     User-Agent: Asterisk PBX
>     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>     Supported: replaces
>     Content-Length: 0
>
>
>     2007-08-15 16:38:17:568 1187188697.568788: Aborting call on unexpected
>     message for Call-Id '[EMAIL PROTECTED]
>     <mailto:[EMAIL PROTECTED]>': whil$
>     Via: SIP/2.0/UDP
>     127.0.0.1:5060
>     <http://127.0.0.1:5060>;branch=z9hG4bK-5633-106-0;received=89.110.157.78
>     <http://89.110.157.78>
>     From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=106
>     To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as6eb78372
>     Call-ID: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>
>     CSeq: 1 INVITE
>     User-Agent: Asterisk PBX
>     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>     Supported: replaces
>     Content-Length: 02007-08-15 16:38:17:568 1187188697.568788: Aborting
>     call on unexpected message for Call-Id '106-5633$
>     Via: SIP/2.0/UDP
>     127.0.0.1:5060
>     <http://127.0.0.1:5060>;branch=z9hG4bK-5633-106-0;received=
>     89.110.157.78 <http://89.110.157.78>
>     From: sipp <sip:[EMAIL PROTECTED]:5060>;tag=106
>     To: sut <sip:[EMAIL PROTECTED]:5060>;tag=as6eb78372
>     Call-ID: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>
>     CSeq: 1 INVITE
>     User-Agent: Asterisk PBX
>     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>     Supported: replaces
>     Content-Length: 0
>
>
>
>
>
>
>     What is going wrong? What can I do to get correct sip calls running?
>     It would be great if someone could help me.
>
>     thanks
>
>     Christian
>
>
>
>
>
>
>
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